diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-04-02 15:50:04 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-04-02 15:50:04 -0700 |
| commit | 848960e576dafc8ed54c691b2f70b92e1fdea9ba (patch) | |
| tree | 27ea80003da03b81f0b188d3712f0194745126d9 /sound | |
| parent | bc3b3f4bfbded031a11c4284106adddbfacd05bb (diff) | |
| parent | 5c6cd7021a05a02fcf37f360592d7c18d4d807fb (diff) | |
| download | linux-stable-848960e576dafc8ed54c691b2f70b92e1fdea9ba.tar.gz linux-stable-848960e576dafc8ed54c691b2f70b92e1fdea9ba.tar.bz2 linux-stable-848960e576dafc8ed54c691b2f70b92e1fdea9ba.zip | |
Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became again a busy development cycle. There are few ALSA core
updates (merely API cleanups and sparse fixes), with the majority of
other changes are found in ASoC scene.
Here are some highlights:
ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)
ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140
HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs
USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus"
* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
ALSA: hda/realtek - a fake key event is triggered by running shutup
ALSA: hda: default enable CA0132 DSP support
ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
ASoC: tlv320adcx140: Remove undocumented property
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: boards: add sof_sdw machine driver
ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
ASoC: rt5682: move DAI clock registry to I2S mode
ASoC: pxa: magician: convert to use i2c_new_client_device()
ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
...
Diffstat (limited to 'sound')
378 files changed, 17505 insertions, 3925 deletions
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index a86c95d89824..e81083e1bc68 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -38,7 +38,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_slave_config config; int ret; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) return 0; @@ -47,7 +47,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, return ret; snd_dmaengine_pcm_set_config_from_dai_data(substream, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream), &config); ret = dmaengine_slave_config(chan, &config); @@ -95,7 +95,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream) runtime->hw = pxa2xx_pcm_hardware; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) return 0; @@ -120,7 +120,7 @@ int pxa2xx_pcm_open(struct snd_pcm_substream *substream) return ret; return snd_dmaengine_pcm_open( - substream, dma_request_slave_channel(rtd->cpu_dai->dev, + substream, dma_request_slave_channel(asoc_rtd_to_cpu(rtd, 0)->dev, dma_params->chan_name)); } EXPORT_SYMBOL(pxa2xx_pcm_open); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 9de1c9a0173e..509290f2efa8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -488,6 +488,48 @@ out: } #endif /* !COMPR_CODEC_CAPS_OVERFLOW */ +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size) +{ + struct snd_dma_buffer *dmab; + int ret; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + dmab = kzalloc(sizeof(*dmab), GFP_KERNEL); + if (!dmab) + return -ENOMEM; + dmab->dev = stream->dma_buffer.dev; + ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab); + if (ret < 0) { + kfree(dmab); + return ret; + } + + snd_compr_set_runtime_buffer(stream, dmab); + stream->runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(snd_compr_malloc_pages); + +int snd_compr_free_pages(struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + if (runtime->dma_area == NULL) + return 0; + if (runtime->dma_buffer_p != &stream->dma_buffer) { + /* It's a newly allocated buffer. Release it now. */ + snd_dma_free_pages(runtime->dma_buffer_p); + kfree(runtime->dma_buffer_p); + } + + snd_compr_set_runtime_buffer(stream, NULL); + return 0; +} +EXPORT_SYMBOL(snd_compr_free_pages); + /* revisit this with snd_pcm_preallocate_xxx */ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) diff --git a/sound/core/device.c b/sound/core/device.c index cdc5af526739..bf0b04a7ee79 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -237,3 +237,24 @@ void snd_device_free_all(struct snd_card *card) list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } + +/** + * snd_device_get_state - Get the current state of the given device + * @card: the card instance + * @device_data: the data pointer to release + * + * Returns the current state of the given device object. For the valid + * device, either @SNDRV_DEV_BUILD, @SNDRV_DEV_REGISTERED or + * @SNDRV_DEV_DISCONNECTED is returned. + * Or for a non-existing device, -1 is returned as an error. + */ +int snd_device_get_state(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + dev = look_for_dev(card, device_data); + if (dev) + return dev->state; + return -1; +} +EXPORT_SYMBOL_GPL(snd_device_get_state); diff --git a/sound/core/info.c b/sound/core/info.c index ca87ae4c30ba..8c6bc5241df5 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -604,7 +604,7 @@ int snd_info_card_free(struct snd_card *card) */ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { - int c = -1; + int c; if (snd_BUG_ON(!buffer || !buffer->buffer)) return 1; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 13db77771f0f..930def8201f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -884,20 +884,17 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) sformat = snd_pcm_plug_slave_format(format, sformat_mask); if ((__force int)sformat < 0 || - !snd_mask_test(sformat_mask, (__force int)sformat)) { - for (sformat = (__force snd_pcm_format_t)0; - (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST; - sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) { - if (snd_mask_test(sformat_mask, (__force int)sformat) && + !snd_mask_test_format(sformat_mask, sformat)) { + pcm_for_each_format(sformat) { + if (snd_mask_test_format(sformat_mask, sformat) && snd_pcm_oss_format_to(sformat) >= 0) - break; - } - if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) { - pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); - err = -EINVAL; - goto failure; + goto format_found; } + pcm_dbg(substream->pcm, "Cannot find a format!!!\n"); + err = -EINVAL; + goto failure; } + format_found: err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0); if (err < 0) goto failure; @@ -1220,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const if (ret < 0) break; } + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret != -EPIPE && ret != -ESTRPIPE) break; /* test, if we can't store new data, because the stream */ @@ -1257,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p ret = snd_pcm_oss_capture_position_fixup(substream, &delay); if (ret < 0) break; + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 752d078908e9..fbda4ebf38b3 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - int stream; + struct snd_pcm_plugin *plugin, *plugin_next; - if (snd_BUG_ON(!plug)) - return -ENXIO; - if (drv_frames == 0) - return 0; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_last(plug); - while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) - drv_frames = plugin->src_frames(plugin, drv_frames); - plugin = plugin_prev; + plugin = snd_pcm_plug_first(plug); + while (plugin && frames > 0) { + plugin_next = plugin->next; + if (plugin->dst_frames) { + frames = plugin->dst_frames(plugin, frames); + if (frames < 0) + return frames; } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_first(plug); - while (plugin && drv_frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) - drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin = plugin_next; + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin = plugin_next; + } + return frames; +} + +static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) +{ + struct snd_pcm_plugin *plugin, *plugin_prev; + + plugin = snd_pcm_plug_last(plug); + while (plugin && frames > 0) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin_prev = plugin->prev; + if (plugin->src_frames) { + frames = plugin->src_frames(plugin, frames); + if (frames < 0) + return frames; } - } else + plugin = plugin_prev; + } + return frames; +} + +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +{ + if (snd_BUG_ON(!plug)) + return -ENXIO; + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_src_frames(plug, drv_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_dst_frames(plug, drv_frames); + default: snd_BUG(); - return drv_frames; + return -EINVAL; + } } snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - snd_pcm_sframes_t frames; - int stream; - if (snd_BUG_ON(!plug)) return -ENXIO; - if (clt_frames == 0) - return 0; - frames = clt_frames; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) { - frames = plugin->dst_frames(plugin, frames); - if (frames < 0) - return frames; - } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin = plugin_next; - } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_last(plug); - while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) { - frames = plugin->src_frames(plugin, frames); - if (frames < 0) - return frames; - } - plugin = plugin_prev; - } - } else + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_dst_frames(plug, clt_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_src_frames(plug, clt_frames); + default: snd_BUG(); - return frames; + return -EINVAL; + } } static int snd_pcm_plug_formats(const struct snd_mask *mask, diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 7cd09cef6961..d381f4c967c9 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -47,7 +47,7 @@ struct rate_priv { unsigned int pos; rate_f func; snd_pcm_sframes_t old_src_frames, old_dst_frames; - struct rate_channel channels[0]; + struct rate_channel channels[]; }; static void rate_init(struct snd_pcm_plugin *plugin) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index a141a301369f..b6d2331a82f7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1019,7 +1019,7 @@ static ssize_t show_pcm_class(struct device *dev, str = "none"; else str = strs[pcm->dev_class]; - return snprintf(buf, PAGE_SIZE, "%s\n", str); + return sprintf(buf, "%s\n", str); } static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 5749a8a49784..4d059ff2b2e4 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -240,6 +240,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; struct dma_tx_state state; enum dma_status status; unsigned int buf_size; @@ -250,9 +251,12 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) buf_size = snd_pcm_lib_buffer_bytes(substream); if (state.residue > 0 && state.residue <= buf_size) pos = buf_size - state.residue; + + runtime->delay = bytes_to_frames(runtime, + state.in_flight_bytes); } - return bytes_to_frames(substream->runtime, pos); + return bytes_to_frames(runtime, pos); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); @@ -426,7 +430,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( * default assumption is that it supports 1, 2 and 4 bytes * widths. */ - for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { + pcm_for_each_format(i) { int bits = snd_pcm_format_physical_width(i); /* diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index a6a541511534..257d412eac5d 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -42,6 +42,11 @@ struct pcm_format_data { /* we do lots of calculations on snd_pcm_format_t; shut up sparse */ #define INT __force int +static bool valid_format(snd_pcm_format_t format) +{ + return (INT)format >= 0 && (INT)format <= (INT)SNDRV_PCM_FORMAT_LAST; +} + static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_S8] = { .width = 8, .phys = 8, .le = -1, .signd = 1, @@ -259,7 +264,7 @@ static const struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = int snd_pcm_format_signed(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].signd) < 0) return -EINVAL; @@ -307,7 +312,7 @@ EXPORT_SYMBOL(snd_pcm_format_linear); int snd_pcm_format_little_endian(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].le) < 0) return -EINVAL; @@ -343,7 +348,7 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian); int snd_pcm_format_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].width) == 0) return -EINVAL; @@ -361,7 +366,7 @@ EXPORT_SYMBOL(snd_pcm_format_width); int snd_pcm_format_physical_width(snd_pcm_format_t format) { int val; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if ((val = pcm_formats[(INT)format].phys) == 0) return -EINVAL; @@ -394,7 +399,7 @@ EXPORT_SYMBOL(snd_pcm_format_size); */ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) { - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return NULL; if (! pcm_formats[(INT)format].phys) return NULL; @@ -418,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int unsigned char *dst; const unsigned char *pat; - if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) + if (!valid_format(format)) return -EINVAL; if (samples == 0) return 0; @@ -474,32 +479,32 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int EXPORT_SYMBOL(snd_pcm_format_set_silence); /** - * snd_pcm_limit_hw_rates - determine rate_min/rate_max fields - * @runtime: the runtime instance + * snd_pcm_hw_limit_rates - determine rate_min/rate_max fields + * @hw: the pcm hw instance * * Determines the rate_min and rate_max fields from the rates bits of - * the given runtime->hw. + * the given hw. * * Return: Zero if successful. */ -int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime) +int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw) { int i; for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_min = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_min = snd_pcm_known_rates.list[i]; break; } } for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { - if (runtime->hw.rates & (1 << i)) { - runtime->hw.rate_max = snd_pcm_known_rates.list[i]; + if (hw->rates & (1 << i)) { + hw->rate_max = snd_pcm_known_rates.list[i]; break; } } return 0; } -EXPORT_SYMBOL(snd_pcm_limit_hw_rates); +EXPORT_SYMBOL(snd_pcm_hw_limit_rates); /** * snd_pcm_rate_to_rate_bit - converts sample rate to SNDRV_PCM_RATE_xxx bit diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index cbdf061667fa..aef860256278 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -228,6 +228,9 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, return err; } +/* macro for simplified cast */ +#define PARAM_MASK_BIT(b) (1U << (__force int)(b)) + static bool hw_support_mmap(struct snd_pcm_substream *substream) { if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) @@ -257,7 +260,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_mask_param_enabled()) @@ -271,7 +274,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_mask_param(substream, k, 0, &old_mask, m); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -293,7 +296,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, return -EINVAL; /* This parameter is not requested to change by a caller. */ - if (!(params->rmask & (1 << k))) + if (!(params->rmask & PARAM_MASK_BIT(k))) continue; if (trace_hw_interval_param_enabled()) @@ -307,7 +310,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream, /* Set corresponding flag so that the caller gets it. */ trace_hw_interval_param(substream, k, 0, &old_interval, i); - params->cmask |= 1 << k; + params->cmask |= PARAM_MASK_BIT(k); } return 0; @@ -349,7 +352,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, * have 0 so that the parameters are never changed anymore. */ for (k = 0; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) - vstamps[k] = (params->rmask & (1 << k)) ? 1 : 0; + vstamps[k] = (params->rmask & PARAM_MASK_BIT(k)) ? 1 : 0; /* Due to the above design, actual sequence number starts at 2. */ stamp = 2; @@ -417,7 +420,7 @@ retry: hw_param_interval(params, r->var)); } - params->cmask |= (1 << r->var); + params->cmask |= PARAM_MASK_BIT(r->var); vstamps[r->var] = stamp; again = true; } @@ -486,9 +489,9 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, params->info = 0; params->fifo_size = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_SAMPLE_BITS)) params->msbits = 0; - if (params->rmask & (1 << SNDRV_PCM_HW_PARAM_RATE)) { + if (params->rmask & PARAM_MASK_BIT(SNDRV_PCM_HW_PARAM_RATE)) { params->rate_num = 0; params->rate_den = 0; } @@ -2293,21 +2296,21 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - unsigned int k; + snd_pcm_format_t k; const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(mask, k)) + if (!snd_mask_test_format(mask, k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) continue; /* ignore invalid formats */ if ((unsigned)bits < i->min || (unsigned)bits > i->max) - snd_mask_reset(&m, k); + snd_mask_reset(&m, (__force unsigned)k); } return snd_mask_refine(mask, &m); } @@ -2316,14 +2319,15 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - unsigned int k; + snd_pcm_format_t k; + t.min = UINT_MAX; t.max = 0; t.openmin = 0; t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + pcm_for_each_format(k) { int bits; - if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) + if (!snd_mask_test_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; bits = snd_pcm_format_physical_width(k); if (bits <= 0) @@ -2505,16 +2509,16 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) unsigned int mask = 0; if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); if (hw_support_mmap(substream)) { if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED); if (hw->info & SNDRV_PCM_INFO_COMPLEX) - mask |= 1 << SNDRV_PCM_ACCESS_MMAP_COMPLEX; + mask |= PARAM_MASK_BIT(SNDRV_PCM_ACCESS_MMAP_COMPLEX); } err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_ACCESS, mask); if (err < 0) @@ -2524,7 +2528,8 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, 1 << SNDRV_PCM_SUBFORMAT_STD); + err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, + PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD)); if (err < 0) return err; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index d78a27271d6d..251eaf1152e2 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -118,7 +118,7 @@ struct loopback_cable { struct loopback_setup { unsigned int notify: 1; unsigned int rate_shift; - unsigned int format; + snd_pcm_format_t format; unsigned int rate; unsigned int channels; struct snd_ctl_elem_id active_id; @@ -1432,7 +1432,7 @@ static int loopback_format_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST; + uinfo->value.integer.max = (__force int)SNDRV_PCM_FORMAT_LAST; uinfo->value.integer.step = 1; return 0; } @@ -1443,7 +1443,7 @@ static int loopback_format_get(struct snd_kcontrol *kcontrol, struct loopback *loopback = snd_kcontrol_chip(kcontrol); ucontrol->value.integer.value[0] = - loopback->setup[kcontrol->id.subdevice] + (__force int)loopback->setup[kcontrol->id.subdevice] [kcontrol->id.device].format; return 0; } diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 02ac3f4e0c02..b5486de08b97 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -901,10 +901,10 @@ static int snd_card_dummy_new_mixer(struct snd_dummy *dummy) static void print_formats(struct snd_dummy *dummy, struct snd_info_buffer *buffer) { - int i; + snd_pcm_format_t i; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy->pcm_hw.formats & (1ULL << i)) + pcm_for_each_format(i) { + if (dummy->pcm_hw.formats & pcm_format_to_bits(i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 976d8cb9a34f..2c8e3392a490 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -509,7 +509,7 @@ MODULE_DEVICE_TABLE(ieee1394, bebob_id_table); static struct fw_driver bebob_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-bebob", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = bebob_probe, diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f5fc0e7c024..c84b913a9fe0 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -192,7 +192,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table); static struct fw_driver dg00x_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-digi00x", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_dg00x_probe, diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index f5a016560eb8..b62a4fd22407 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_ff_id_table); static struct fw_driver ff_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireface", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_ff_probe, diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 134fc9ee26b9..b1cc013a3540 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -362,7 +362,7 @@ MODULE_DEVICE_TABLE(ieee1394, efw_id_table); static struct fw_driver efw_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-fireworks", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = efw_probe, diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index c29a97f6f638..6f38335fe10b 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -17,6 +17,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, long count, loff_t *offset) + __releases(&tscm->lock) { struct snd_firewire_event_lock_status event = { .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, @@ -36,6 +37,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, long remained, loff_t *offset) + __releases(&tscm->lock) { char __user *pos = buf; unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index addc464503bc..5dac0d9fc58e 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -224,7 +224,7 @@ MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table); static struct fw_driver tscm_driver = { .driver = { .owner = THIS_MODULE, - .name = "snd-firewire-tascam", + .name = KBUILD_MODNAME, .bus = &fw_bus_type, }, .probe = snd_tscm_probe, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 9a526aeef8da..e3119f5cb0d5 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -204,7 +204,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); */ int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) { - return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + return scnprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", codec->vendor_id, codec->revision_id, codec->type); } EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index e377ac93f37f..8e8257c574b0 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -435,7 +435,7 @@ enum { #define LOOP_WRITE(rec, offset, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = (rec)->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, offset); \ while (count > 0) { \ @@ -492,7 +492,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs, #define LOOP_WRITE(rec, pos, _buf, count, mode) \ do { \ struct snd_emu8000 *emu = rec->emu; \ - unsigned short *buf = (unsigned short *)(_buf); \ + unsigned short *buf = (__force unsigned short *)(_buf); \ snd_emu8000_write_wait(emu, 1); \ EMU8000_SMALW_WRITE(emu, pos + rec->loop_start[0]); \ if (rec->voices > 1) \ diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4f524a9dbbca..4462375d2d82 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1070,7 +1070,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, { struct snd_ali *codec = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; - unsigned int what, whati, capture_flag; + unsigned int what, whati; struct snd_ali_voice *pvoice, *evoice; unsigned int val; int do_start; @@ -1088,7 +1088,7 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, return -EINVAL; } - what = whati = capture_flag = 0; + what = whati = 0; snd_pcm_group_for_each_entry(s, substream) { if ((struct snd_ali *) snd_pcm_substream_chip(s) == codec) { pvoice = s->runtime->private_data; @@ -1110,8 +1110,6 @@ static int snd_ali_trigger(struct snd_pcm_substream *substream, evoice->running = 0; } snd_pcm_trigger_done(s, substream); - if (pvoice->mode) - capture_flag = 1; } } spin_lock(&codec->reg_lock); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a89a7e603ca8..6ff581733a19 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1789,6 +1789,7 @@ int snd_emu10k1_create(struct snd_card *card, int idx, err; int is_audigy; size_t page_table_size; + __le32 *pgtbl; unsigned int silent_page; const struct snd_emu_chip_details *c; static const struct snd_device_ops ops = { @@ -2009,8 +2010,9 @@ int snd_emu10k1_create(struct snd_card *card, /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, emu->silent_page.bytes); silent_page = emu->silent_page.addr << emu->address_mode; + pgtbl = (__le32 *)emu->ptb_pages.area; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) - ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); + pgtbl[idx] = cpu_to_le32(silent_page | idx); /* set up voice indices */ for (idx = 0; idx < NUM_G; idx++) { diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 53e7732ef752..a34a2c9f4bcf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -88,7 +88,7 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; - hda_nid_t conns[0]; + hda_nid_t conns[]; }; /* look up the cached results */ diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2609e391ce54..9765652a73d7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -373,7 +373,7 @@ static int azx_get_sync_time(ktime_t *device, u32 wallclk_ctr, wallclk_cycles; bool direction; u32 dma_select; - u32 timeout = 200; + u32 timeout; u32 retry_count = 0; runtime = substream->runtime; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ded8bc07d755..34fe753a46fb 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1180,6 +1180,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), {} @@ -2698,7 +2699,7 @@ struct dsp_image_seg { u32 magic; u32 chip_addr; u32 count; - u32 data[0]; + u32 data[]; }; static const u32 g_magic_value = 0x4c46584d; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5119a9ae3d8a..bb287a916dae 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -154,7 +154,6 @@ struct hdmi_spec { struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; - bool use_jack_detect; /* jack detection enabled */ bool use_acomp_notifier; /* use eld_notify callback for hotplug */ bool acomp_registered; /* audio component registered in this driver */ struct drm_audio_component_audio_ops drm_audio_ops; @@ -753,7 +752,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, * Unsolicited events */ -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, int dev_id) @@ -764,8 +763,7 @@ static void check_presence_and_report(struct hda_codec *codec, hda_nid_t nid, if (pin_idx < 0) return; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + hdmi_present_sense(get_pin(spec, pin_idx), 1); mutex_unlock(&spec->pcm_lock); } @@ -779,21 +777,9 @@ static void jack_callback(struct hda_codec *codec, check_presence_and_report(codec, jack->nid, jack->dev_id); } -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res, + struct hda_jack_tbl *jack) { - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - struct hda_jack_tbl *jack; - - if (codec->dp_mst) { - int dev_entry = - (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; - - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); - } else { - jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); - } - if (!jack) - return; jack->jack_dirty = 1; codec_dbg(codec, @@ -853,7 +839,7 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) } if (subtag == 0) - hdmi_intrinsic_event(codec, res); + hdmi_intrinsic_event(codec, res, jack); else hdmi_non_intrinsic_event(codec, res); } @@ -1480,21 +1466,60 @@ static void hdmi_pcm_reset_pin(struct hdmi_spec *spec, per_pin->channels = 0; } +static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + struct hdmi_spec *spec = codec->spec; + + if (per_pin->pcm_idx >= 0) + return spec->pcm_rec[per_pin->pcm_idx].jack; + else + return NULL; +} + /* update per_pin ELD from the given new ELD; * setup info frame and notification accordingly + * also notify ELD kctl and report jack status changes */ -static bool update_eld(struct hda_codec *codec, +static void update_eld(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, - struct hdmi_eld *eld) + struct hdmi_eld *eld, + int repoll) { struct hdmi_eld *pin_eld = &per_pin->sink_eld; struct hdmi_spec *spec = codec->spec; + struct snd_jack *pcm_jack; bool old_eld_valid = pin_eld->eld_valid; bool eld_changed; int pcm_idx; + if (eld->eld_valid) { + if (eld->eld_size <= 0 || + snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, + eld->eld_size) < 0) { + eld->eld_valid = false; + if (repoll) { + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); + return; + } + } + } + + if (!eld->eld_valid || eld->eld_size <= 0) { + eld->eld_valid = false; + eld->eld_size = 0; + } + /* for monitor disconnection, save pcm_idx firstly */ pcm_idx = per_pin->pcm_idx; + + /* + * pcm_idx >=0 before update_eld() means it is in monitor + * disconnected event. Jack must be fetched before update_eld(). + */ + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); + if (spec->dyn_pcm_assign) { if (eld->eld_valid) { hdmi_attach_hda_pcm(spec, per_pin); @@ -1509,6 +1534,8 @@ static bool update_eld(struct hda_codec *codec, */ if (pcm_idx == -1) pcm_idx = per_pin->pcm_idx; + if (!pcm_jack) + pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); if (eld->eld_valid) snd_hdmi_show_eld(codec, &eld->info); @@ -1547,42 +1574,17 @@ static bool update_eld(struct hda_codec *codec, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &get_hdmi_pcm(spec, pcm_idx)->eld_ctl->id); - return eld_changed; -} -static struct snd_jack *pin_idx_to_pcm_jack(struct hda_codec *codec, - struct hdmi_spec_per_pin *per_pin) -{ - struct hdmi_spec *spec = codec->spec; - struct snd_jack *jack = NULL; - struct hda_jack_tbl *jack_tbl; - - /* if !dyn_pcm_assign, get jack from hda_jack_tbl - * in !dyn_pcm_assign case, spec->pcm_rec[].jack is not - * NULL even after snd_hda_jack_tbl_clear() is called to - * free snd_jack. This may cause access invalid memory - * when calling snd_jack_report - */ - if (per_pin->pcm_idx >= 0 && spec->dyn_pcm_assign) { - jack = spec->pcm_rec[per_pin->pcm_idx].jack; - } else if (!spec->dyn_pcm_assign) { - /* - * jack tbl doesn't support DP MST - * DP MST will use dyn_pcm_assign, - * so DP MST will never come here - */ - jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack_tbl) - jack = jack_tbl->jack; - } - return jack; + if (eld_changed && pcm_jack) + snd_jack_report(pcm_jack, + (eld->monitor_present && eld->eld_valid) ? + SND_JACK_AVOUT : 0); } + /* update ELD and jack state via HD-audio verbs */ -static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, +static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, int repoll) { - struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; @@ -1597,9 +1599,11 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, * the unsolicited response to avoid custom WARs. */ int present; - bool ret; - bool do_repoll = false; - struct snd_jack *pcm_jack = NULL; + int ret; + + ret = snd_hda_power_up_pm(codec); + if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) + goto out; present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); @@ -1618,62 +1622,12 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, if (spec->ops.pin_get_eld(codec, pin_nid, dev_id, eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; - else { - if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, - eld->eld_size) < 0) - eld->eld_valid = false; - } - if (!eld->eld_valid && repoll) - do_repoll = true; } - if (do_repoll) { - schedule_delayed_work(&per_pin->work, msecs_to_jiffies(300)); - } else { - /* - * pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before - * update_eld(). - */ - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - update_eld(codec, per_pin, eld); - if (!pcm_jack) - pcm_jack = pin_idx_to_pcm_jack(codec, per_pin); - } - - ret = !repoll || !eld->monitor_present || eld->eld_valid; - - jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id); - if (jack) { - jack->block_report = !ret; - jack->pin_sense = (eld->monitor_present && eld->eld_valid) ? - AC_PINSENSE_PRESENCE : 0; - - if (spec->dyn_pcm_assign && pcm_jack && !do_repoll) { - int state = 0; - - if (jack->pin_sense & AC_PINSENSE_PRESENCE) - state = SND_JACK_AVOUT; - snd_jack_report(pcm_jack, state); - } - - /* - * snd_hda_jack_pin_sense() call at the beginning of this - * function, updates jack->pins_sense and clears - * jack->jack_dirty, therefore snd_hda_jack_report_sync() will - * not override the jack->pin_sense. - * - * snd_hda_jack_report_sync() is superfluous for dyn_pcm_assign - * case. The jack->pin_sense update was already performed, and - * hda_jack->jack is NULL for dyn_pcm_assign. - * - * Don't call snd_hda_jack_report_sync() for - * dyn_pcm_assign. - */ - ret = ret && !spec->dyn_pcm_assign; - } + update_eld(codec, per_pin, eld, repoll); mutex_unlock(&per_pin->lock); - return ret; + out: + snd_hda_power_down_pm(codec); } /* update ELD and jack state via audio component */ @@ -1682,64 +1636,25 @@ static void sync_eld_via_acomp(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; - struct snd_jack *jack = NULL; - bool changed; - int size; mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); - if (size > 0) { - size = min(size, ELD_MAX_SIZE); - if (snd_hdmi_parse_eld(codec, &eld->info, - eld->eld_buffer, size) < 0) - size = -EINVAL; - } - - if (size > 0) { - eld->eld_valid = true; - eld->eld_size = size; - } else { - eld->eld_valid = false; - eld->eld_size = 0; - } - - /* pcm_idx >=0 before update_eld() means it is in monitor - * disconnected event. Jack must be fetched before update_eld() - */ - jack = pin_idx_to_pcm_jack(codec, per_pin); - changed = update_eld(codec, per_pin, eld); - if (jack == NULL) - jack = pin_idx_to_pcm_jack(codec, per_pin); - if (changed && jack) - snd_jack_report(jack, - (eld->monitor_present && eld->eld_valid) ? - SND_JACK_AVOUT : 0); + eld->eld_valid = (eld->eld_size > 0); + update_eld(codec, per_pin, eld, 0); mutex_unlock(&per_pin->lock); } -static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; - int ret; - /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) { - ret = snd_hda_power_up_pm(codec); - if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { - snd_hda_power_down_pm(codec); - return false; - } - ret = hdmi_present_sense_via_verbs(per_pin, repoll); - snd_hda_power_down_pm(codec); - } else { + if (!codec_has_acomp(codec)) + hdmi_present_sense_via_verbs(per_pin, repoll); + else sync_eld_via_acomp(codec, per_pin); - ret = false; /* don't call snd_hda_jack_report_sync() */ - } - - return ret; } static void hdmi_repoll_eld(struct work_struct *work) @@ -1759,8 +1674,7 @@ static void hdmi_repoll_eld(struct work_struct *work) per_pin->repoll_count = 0; mutex_lock(&spec->pcm_lock); - if (hdmi_present_sense(per_pin, per_pin->repoll_count)) - snd_hda_jack_report_sync(per_pin->codec); + hdmi_present_sense(per_pin, per_pin->repoll_count); mutex_unlock(&spec->pcm_lock); } @@ -2206,15 +2120,23 @@ static void free_hdmi_jack_priv(struct snd_jack *jack) pcm->jack = NULL; } -static int add_hdmi_jack_kctl(struct hda_codec *codec, - struct hdmi_spec *spec, - int pcm_idx, - const char *name) +static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) { + char hdmi_str[32] = "HDMI/DP"; + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pcm_idx); struct snd_jack *jack; + int pcmdev = get_pcm_rec(spec, pcm_idx)->device; int err; - err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack, + if (pcmdev > 0) + sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); + if (!spec->dyn_pcm_assign && + !is_jack_detectable(codec, per_pin->pin_nid)) + strncat(hdmi_str, " Phantom", + sizeof(hdmi_str) - strlen(hdmi_str) - 1); + + err = snd_jack_new(codec->card, hdmi_str, SND_JACK_AVOUT, &jack, true, false); if (err < 0) return err; @@ -2225,48 +2147,6 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec, return 0; } -static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) -{ - char hdmi_str[32] = "HDMI/DP"; - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - struct hda_jack_tbl *jack; - int pcmdev = get_pcm_rec(spec, pcm_idx)->device; - bool phantom_jack; - int ret; - - if (pcmdev > 0) - sprintf(hdmi_str + strlen(hdmi_str), ",pcm=%d", pcmdev); - - if (spec->dyn_pcm_assign) - return add_hdmi_jack_kctl(codec, spec, pcm_idx, hdmi_str); - - /* for !dyn_pcm_assign, we still use hda_jack for compatibility */ - /* if !dyn_pcm_assign, it must be non-MST mode. - * This means pcms and pins are statically mapped. - * And pcm_idx is pin_idx. - */ - per_pin = get_pin(spec, pcm_idx); - phantom_jack = !is_jack_detectable(codec, per_pin->pin_nid); - if (phantom_jack) - strncat(hdmi_str, " Phantom", - sizeof(hdmi_str) - strlen(hdmi_str) - 1); - ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid, - per_pin->dev_id, hdmi_str, phantom_jack, - 0, NULL); - if (ret < 0) - return ret; - jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, - per_pin->dev_id); - if (jack == NULL) - return 0; - /* assign jack->jack to pcm_rec[].jack to - * align with dyn_pcm_assign mode - */ - spec->pcm_rec[pcm_idx].jack = jack->jack; - return 0; -} - static int generic_hdmi_build_controls(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2355,7 +2235,6 @@ static int generic_hdmi_init(struct hda_codec *codec) int pin_idx; mutex_lock(&spec->bind_lock); - spec->use_jack_detect = !codec->jackpoll_interval; for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; @@ -2365,12 +2244,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); if (codec_has_acomp(codec)) continue; - if (spec->use_jack_detect) - snd_hda_jack_detect_enable(codec, pin_nid, dev_id); - else - snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, - dev_id, - jack_callback); + snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, dev_id, + jack_callback); } mutex_unlock(&spec->bind_lock); return 0; @@ -2532,12 +2407,6 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int val = use_acomp ? 0 : (AC_USRSP_EN | tbl->tag); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, val); - } else { - /* if no jack entry was defined beforehand, create a new one - * at need (i.e. only when notifier is cleared) - */ - if (!use_acomp) - snd_hda_jack_detect_enable(codec, nid, dev_id); } } @@ -2553,13 +2422,11 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, spec->use_acomp_notifier = use_acomp; spec->codec->relaxed_resume = use_acomp; /* reprogram each jack detection logic depending on the notifier */ - if (spec->use_jack_detect) { - for (i = 0; i < spec->num_pins; i++) - reprogram_jack_detect(spec->codec, - get_pin(spec, i)->pin_nid, - get_pin(spec, i)->dev_id, - use_acomp); - } + for (i = 0; i < spec->num_pins; i++) + reprogram_jack_detect(spec->codec, + get_pin(spec, i)->pin_nid, + get_pin(spec, i)->dev_id, + use_acomp); mutex_unlock(&spec->bind_lock); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b..f66a48154a57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -367,7 +368,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: + case 0x10ec0236: case 0x10ec0255: + case 0x10ec0256: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -379,11 +382,6 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x36, 0x5757); - alc_update_coef_idx(codec, 0x10, 1<<9, 0); - break; case 0x10ec0275: alc_update_coef_idx(codec, 0xe, 0, 1<<0); break; @@ -2982,6 +2980,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3269,7 +3368,13 @@ static void alc256_init(struct hda_codec *codec) alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */ alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15); - alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ + /* + * Expose headphone mic (or possibly Line In on some machines) instead + * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See + * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of + * this register. + */ + alc_write_coef_idx(codec, 0x36, 0x5757); } static void alc256_shutup(struct hda_codec *codec) @@ -3372,6 +3477,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3518,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -4008,6 +4118,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10); } +static void alc285_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00); +} + static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5375,17 +5491,6 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } -static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, - const struct hda_fixup *fix, - int action) -{ - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - - snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); - snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); -} - static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5662,69 +5767,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5863,8 +5905,6 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, - ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, - ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -5923,6 +5963,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC285_FIXUP_HP_GPIO_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -6604,23 +6645,6 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, - [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = { - .type = HDA_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* Disable pass-through path for FRONT 14h */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x36}, - {0x20, AC_VERB_SET_PROC_COEF, 0x1737}, - {} - }, - .chained = true, - .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE - }, - [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc256_fixup_dell_xps_13_headphone_noise2, - .chained = true, - .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE - }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7061,6 +7085,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC285_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7114,17 +7142,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), @@ -7208,6 +7233,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7477,7 +7503,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"}, {.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"}, {.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"}, - {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"}, {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 21ab9cc50c71..65a887b217ee 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -30,7 +30,7 @@ #if K1212_DEBUG_LEVEL > 0 #define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) #else -#define K1212_DEBUG_PRINTK(fmt,...) +#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0) #endif #if K1212_DEBUG_LEVEL > 1 #define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cc06f0a1a7e4..227aece17e39 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) return; } } else { - int err = -EINVAL; + int err; + err = hdsp_request_fw_loader(hdsp); if (err < 0) { snd_iprintf(buffer, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 799789c8eea9..8b03e2dc503f 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -414,6 +414,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -435,6 +436,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -451,7 +453,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = snd_pcm_sgbuf_get_chunk_size(substream, ofs, rest); rest -= r; if (! rest) { @@ -466,7 +468,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 84e589803e2e..607b7100db1c 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -267,6 +267,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre { unsigned int i, idx, ofs, rest; struct via82xx_modem *chip = snd_pcm_substream_chip(substream); + __le32 *pgtbl; if (dev->table.area == NULL) { /* the start of each lists must be aligned to 8 bytes, @@ -288,6 +289,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* fill the entries */ idx = 0; ofs = 0; + pgtbl = (__le32 *)dev->table.area; for (i = 0; i < periods; i++) { rest = fragsize; /* fill descriptors for a period. @@ -304,7 +306,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre return -EINVAL; } addr = snd_pcm_sgbuf_get_addr(substream, ofs); - ((u32 *)dev->table.area)[idx << 1] = cpu_to_le32(addr); + pgtbl[idx << 1] = cpu_to_le32(addr); r = PAGE_SIZE - (ofs % PAGE_SIZE); if (rest < r) r = rest; @@ -321,7 +323,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre "tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); */ - ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); + pgtbl[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; ofs += r; diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 093806d735c6..9554a0c506af 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -40,6 +40,7 @@ static int keywest_probe(struct i2c_client *client, static int keywest_attach_adapter(struct i2c_adapter *adapter) { struct i2c_board_info info; + struct i2c_client *client; if (! keywest_ctx) return -EINVAL; @@ -50,9 +51,11 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) memset(&info, 0, sizeof(struct i2c_board_info)); strlcpy(info.type, "keywest", I2C_NAME_SIZE); info.addr = keywest_ctx->addr; - keywest_ctx->client = i2c_new_device(adapter, &info); - if (!keywest_ctx->client) - return -ENODEV; + client = i2c_new_client_device(adapter, &info); + if (IS_ERR(client)) + return PTR_ERR(client); + keywest_ctx->client = client; + /* * We know the driver is already loaded, so the device should be * already bound. If not it means binding failed, and then there diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f40517717c4..bce4cee5cb54 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -26,3 +26,13 @@ config SND_SOC_AMD_ACP3x depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform + +config SND_SOC_AMD_RV_RT5682_MACH + tristate "AMD RV support for RT5682" + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_CROS_EC_CODEC + select I2C_CROS_EC_TUNNEL + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC + help + This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index c4ddc6adb6f0..e6f3d9b469f3 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -2,8 +2,10 @@ acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o +snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ +obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 7a5621e5e233..9414d7269c4f 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -54,7 +54,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 91abeb92b648..73b31f88a6b5 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -48,7 +48,7 @@ static int cz_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, CZ_PLAT_CLK, params_rate(params) * 512); @@ -73,7 +73,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card; struct snd_soc_component *codec; - codec = rtd->codec_dai->component; + codec = asoc_rtd_to_codec(rtd, 0)->component; card = rtd->card; ret = snd_soc_card_jack_new(card, "Headset Jack", diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c new file mode 100644 index 000000000000..024a7ee54cd5 --- /dev/null +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -0,0 +1,376 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec. +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/io.h> +#include <linux/acpi.h> + +#include "raven/acp3x.h" +#include "../codecs/rt5682.h" + +#define PCO_PLAT_CLK 48000000 +#define RT5682_PLL_FREQ (48000 * 512) +#define DUAL_CHANNEL 2 + +static struct snd_soc_jack pco_jack; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; +static struct gpio_desc *dmic_sel; + +static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); + + /* set rt5682 dai fmt */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + dev_err(rtd->card->dev, + "Failed to set rt5682 dai fmt: %d\n", ret); + return ret; + } + + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + PCO_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + + /* Set tdm/i2s1 master bclk ratio */ + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 tdm bclk ratio: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk"); + rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pco_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &pco_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* RT5682 will support only 48K output with 48M mclk */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int acp3x_5682_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 0); + + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 1); + + return rt5682_clk_enable(substream); +} + +static void rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + +static const struct snd_soc_ops acp3x_5682_ops = { + .startup = acp3x_5682_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_max_play_ops = { + .startup = acp3x_max_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap0_ops = { + .startup = acp3x_ec_dmic0_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap1_ops = { + .startup = acp3x_ec_dmic1_startup, + .shutdown = rt5682_shutdown, +}; + +SND_SOC_DAILINK_DEF(acp3x_i2s, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); +SND_SOC_DAILINK_DEF(acp3x_bt, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2"))); + +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); +SND_SOC_DAILINK_DEF(max, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(cros_ec, + DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); + +static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { + { + .name = "acp3x-5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = acp3x_5682_init, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &acp3x_5682_ops, + SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), + }, + { + .name = "acp3x-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &acp3x_max_play_ops, + SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + }, + { + .name = "acp3x-ec-dmic0-capture", + .stream_name = "Capture DMIC0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap0_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, + { + .name = "acp3x-ec-dmic1-capture", + .stream_name = "Capture DMIC1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap1_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, +}; + +static const struct snd_soc_dapm_widget acp3x_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_audio_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_card = { + .name = "acp3xalc5682m98357", + .owner = THIS_MODULE, + .dai_link = acp3x_dai_5682_98357, + .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dapm_widgets = acp3x_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), + .dapm_routes = acp3x_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), + .controls = acp3x_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_controls), +}; + +static int acp3x_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + struct acp3x_platform_info *machine; + + machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card = &acp3x_card; + acp3x_card.dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); + if (IS_ERR(dmic_sel)) { + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", + PTR_ERR(dmic_sel)); + return PTR_ERR(dmic_sel); + } + + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + if (ret) { + dev_err(&pdev->dev, + "devm_snd_soc_register_card(%s) failed: %d\n", + acp3x_card.name, ret); + return ret; + } + return 0; +} + +static const struct acpi_device_id acp3x_audio_acpi_match[] = { + { "AMDI5682", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); + +static struct platform_driver acp3x_audio = { + .driver = { + .name = "acp3x-alc5682-max98357", + .acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = acp3x_probe, +}; + +module_platform_driver(acp3x_audio); + +MODULE_AUTHOR("akshu.agrawal@amd.com"); +MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 91a388184e52..3a3c47e820ab 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -42,7 +42,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, u32 rx_mask, int slots, int slot_width) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_reg, frm_len; + u32 frm_len; u16 slot_len; adata = snd_soc_dai_get_drvdata(cpu_dai); @@ -64,36 +64,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - - /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/ - frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); - if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_ITER; - frmt_reg = mmACP_BTTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_ITER; - frmt_reg = mmACP_I2STDM_TXFRMT; - } - } else { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_IRER; - frmt_reg = mmACP_BTTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_IRER; - frmt_reg = mmACP_I2STDM_RXFRMT; - } - } - val = rv_readl(adata->acp3x_base + reg_val); - rv_writel(val | 0x2, adata->acp3x_base + reg_val); - rv_writel(frm_len, adata->acp3x_base + frmt_reg); adata->tdm_fmt = frm_len; return 0; } @@ -105,12 +76,14 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; struct acp3x_platform_info *pinfo; + struct i2s_dev_data *adata; u32 val; - u32 reg_val; + u32 reg_val, frmt_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; card = prtd->card; + adata = snd_soc_dai_get_drvdata(dai); pinfo = snd_soc_card_get_drvdata(card); if (pinfo) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -141,21 +114,30 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_ITER; + frmt_reg = mmACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_ITER; + frmt_reg = mmACP_I2STDM_TXFRMT; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_IRER; + frmt_reg = mmACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_IRER; + frmt_reg = mmACP_I2STDM_RXFRMT; } } + if (adata->tdm_mode) { + val = rv_readl(rtd->acp3x_base + reg_val); + rv_writel(val | 0x2, rtd->acp3x_base + reg_val); + rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); + } val = rv_readl(rtd->acp3x_base + reg_val); val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e..e362f0bc9e46 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index da60e2ec5535..f25ce50f1a90 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -38,8 +38,13 @@ static int acp3x_power_on(void __iomem *acp3x_base) timeout = 0; while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if (!val) + if (!val) { + /* Set PME_EN as after ACP power On, + * PME_EN gets cleared + */ + rv_writel(0x1, acp3x_base + mmACP_PME_EN); return 0; + } udelay(1); } return -ETIMEDOUT; diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index db67f5ba1e9a..cb03c4f7324c 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -56,7 +56,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pcm_dma_params *prtd; - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (ssc_sr & prtd->mask->ssc_error) { if (snd_pcm_running(substream)) @@ -83,7 +83,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, struct ssc_device *ssc; int ret; - prtd = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); ssc = prtd->ssc; ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 59c1331a6984..a8daebcbf6c8 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -213,7 +213,7 @@ static int atmel_pcm_hw_params(struct snd_soc_component *component, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + prtd->params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 776b27d3686e..148c943cb538 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -27,7 +27,7 @@ static int atmel_asoc_wm8904_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_pll(codec_dai, WM8904_FLL_MCLK, WM8904_FLL_MCLK, diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index befc2a3a05b0..3cb63886195f 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -239,10 +239,10 @@ struct mchp_i2s_mcc_dev { unsigned int frame_length; int tdm_slots; int channels; - int gclk_use:1; - int gclk_running:1; - int tx_rdy:1; - int rx_rdy:1; + unsigned int gclk_use:1; + unsigned int gclk_running:1; + unsigned int tx_rdy:1; + unsigned int rx_rdy:1; }; static irqreturn_t mchp_i2s_mcc_interrupt(int irq, void *dev_id) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index aa6d0d78566f..f9a85fd01b79 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -21,7 +21,7 @@ static int snd_proto_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set proto sysclk */ int ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index b1bef2bf142d..ed1f69b57024 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -96,7 +96,7 @@ static const struct snd_soc_dapm_route intercon[] = { */ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct device *dev = rtd->dev; int ret; diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7822425d5e61..9fbc3c1113cc 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -40,7 +40,7 @@ struct sam9x5_drvdata { */ static int sam9x5_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct device *dev = rtd->dev; int ret; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index d6b692fff29a..d649037bda9b 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -95,7 +95,7 @@ static struct snd_soc_card db1550_ac97_machine = { static int db1200_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* WM8731 has its own 12MHz crystal */ snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 8f855644c6b4..e82bbf2d1eea 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -281,7 +281,7 @@ static int au1xpsc_pcm_open(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = substream->private_data; int stype = substream->stream, *dmaids; - dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dmaids) return -ENODEV; /* whoa, has ordering changed? */ diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index c9a038a5e2d3..4e246c7e78f2 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -195,7 +195,7 @@ static int alchemy_pcm_open(struct snd_soc_component *component, int *dmaids, s = substream->stream; char *name; - dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dmaids = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dmaids) return -ENODEV; /* whoa, has ordering changed? */ diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 0227993c5da8..05eb36991f14 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -58,7 +58,7 @@ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) { struct snd_soc_card *c = x->bus->card->private_data; - return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); + return snd_soc_dai_get_drvdata(c->asoc_rtd_to_cpu(rtd, 0)); } #else diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 0037e96aa228..4218057b0874 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -17,3 +17,12 @@ config SND_SOC_CYGNUS Cygnus chips (bcm958300, bcm958305, bcm911360) If you don't know what to do here, say N. + +config SND_BCM63XX_I2S_WHISTLER + tristate "SoC Audio support for the Broadcom BCM63XX I2S module" + select REGMAP_MMIO + help + Say Y if you want to add support for ASoC audio on Broadcom + DSL/PON chips (bcm63158, bcm63178) + + If you don't know what to do here, say N diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index b81fa421ec27..7c2d7899603b 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -9,3 +9,7 @@ snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o +# BCM63XX Platform Support +snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o + +obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o
\ No newline at end of file diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c new file mode 100644 index 000000000000..246a57ac6679 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -0,0 +1,317 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/bcm/bcm63xx-i2s-whistler.c +// BCM63xx whistler i2s driver +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com> + +#include <linux/clk.h> +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "bcm63xx-i2s.h" + +#define DRV_NAME "brcm-i2s" + +static bool brcm_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_TX_DESC_IFF_LEN: + case I2S_TX_CFG_2 ... I2S_RX_DESC_IFF_LEN: + case I2S_RX_CFG_2 ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG: + case I2S_TX_IRQ_CTL: + case I2S_TX_DESC_IFF_ADDR: + case I2S_TX_DESC_IFF_LEN: + case I2S_TX_DESC_OFF_ADDR: + case I2S_TX_DESC_OFF_LEN: + case I2S_TX_CFG_2: + case I2S_RX_CFG: + case I2S_RX_IRQ_CTL: + case I2S_RX_DESC_OFF_ADDR: + case I2S_RX_DESC_OFF_LEN: + case I2S_RX_DESC_IFF_LEN: + case I2S_RX_DESC_IFF_ADDR: + case I2S_RX_CFG_2: + return true; + default: + return false; + } +} + +static const struct regmap_config brcm_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_REG_MAX, + .writeable_reg = brcm_i2s_wr_reg, + .readable_reg = brcm_i2s_rd_reg, + .volatile_reg = brcm_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int bcm63xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + + ret = clk_set_rate(i2s_priv->i2s_clk, params_rate(params)); + if (ret < 0) + dev_err(i2s_priv->dev, + "Can't set sample rate, err: %d\n", ret); + + return ret; +} + +static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int slavemode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 1); + + /* TX and RX block each have an independent bit to indicate + * if it is generating the clock for the I2S bus. The bus + * clocks need to be generated from either the TX or RX block, + * but not both + */ + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode); + if (slavemode & I2S_RX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + else + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 1); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode); + if (slavemode & I2S_TX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, 0); + else + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_SLAVE_MODE); + } + return 0; +} + +static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int enabled, slavemode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode); + slavemode = slavemode & I2S_TX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_RX_CFG, &enabled); + enabled = enabled & I2S_RX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_MASTER_MODE); + } + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode); + slavemode = slavemode & I2S_RX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_TX_CFG, &enabled); + enabled = enabled & I2S_TX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + } + + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, I2S_RX_SLAVE_MODE); + } +} + +static const struct snd_soc_dai_ops bcm63xx_i2s_dai_ops = { + .startup = bcm63xx_i2s_startup, + .shutdown = bcm63xx_i2s_shutdown, + .hw_params = bcm63xx_i2s_hw_params, +}; + +static struct snd_soc_dai_driver bcm63xx_i2s_dai = { + .name = DRV_NAME, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &bcm63xx_i2s_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, +}; + +static const struct snd_soc_component_driver bcm63xx_i2s_component = { + .name = "bcm63xx", +}; + +static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) +{ + int ret = 0; + void __iomem *regs; + struct resource *r_mem, *region; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + + i2s_priv = devm_kzalloc(&pdev->dev, sizeof(*i2s_priv), GFP_KERNEL); + if (!i2s_priv) + return -ENOMEM; + + i2s_clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(i2s_clk)) { + dev_err(&pdev->dev, "%s: cannot get a brcm clock: %ld\n", + __func__, PTR_ERR(i2s_clk)); + return PTR_ERR(i2s_clk); + } + + r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r_mem) { + dev_err(&pdev->dev, "Unable to get register resource.\n"); + return -ENODEV; + } + + region = devm_request_mem_region(&pdev->dev, r_mem->start, + resource_size(r_mem), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + return -EBUSY; + } + + regs = devm_ioremap_resource(&pdev->dev, r_mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + return ret; + } + + regmap_i2s = devm_regmap_init_mmio(&pdev->dev, + regs, &brcm_i2s_regmap_config); + if (IS_ERR(regmap_i2s)) + return PTR_ERR(regmap_i2s); + + regmap_update_bits(regmap_i2s, I2S_MISC_CFG, + I2S_PAD_LVL_LOOP_DIS_MASK, + I2S_PAD_LVL_LOOP_DIS_ENABLE); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_i2s_component, + &bcm63xx_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register the dai\n"); + return ret; + } + + i2s_priv->dev = &pdev->dev; + i2s_priv->i2s_clk = i2s_clk; + i2s_priv->regmap_i2s = regmap_i2s; + dev_set_drvdata(&pdev->dev, i2s_priv); + + ret = bcm63xx_soc_platform_probe(pdev, i2s_priv); + if (ret) + dev_err(&pdev->dev, "failed to register the pcm\n"); + + return ret; +} + +static int bcm63xx_i2s_dev_remove(struct platform_device *pdev) +{ + bcm63xx_soc_platform_remove(pdev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id snd_soc_bcm_audio_match[] = { + {.compatible = "brcm,bcm63xx-i2s"}, + { } +}; +#endif + +static struct platform_driver bcm63xx_i2s_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), + }, + .probe = bcm63xx_i2s_dev_probe, + .remove = bcm63xx_i2s_dev_remove, +}; + +module_platform_driver(bcm63xx_i2s_driver); + +MODULE_AUTHOR("Kevin,Li <kevin-ke.li@broadcom.com>"); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC I2S Interface"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/bcm/bcm63xx-i2s.h b/sound/soc/bcm/bcm63xx-i2s.h new file mode 100644 index 000000000000..edc328ba53d3 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s.h @@ -0,0 +1,90 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/soc/bcm/bcm63xx-i2s.h +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com> + +#ifndef __BCM63XX_I2S_H +#define __BCM63XX_I2S_H + +#define I2S_DESC_FIFO_DEPTH 8 +#define I2S_MISC_CFG (0x003C) +#define I2S_PAD_LVL_LOOP_DIS_MASK (1 << 2) +#define I2S_PAD_LVL_LOOP_DIS_ENABLE I2S_PAD_LVL_LOOP_DIS_MASK + +#define I2S_TX_ENABLE_MASK (1 << 31) +#define I2S_TX_ENABLE I2S_TX_ENABLE_MASK +#define I2S_TX_OUT_R (1 << 19) +#define I2S_TX_DATA_ALIGNMENT (1 << 2) +#define I2S_TX_DATA_ENABLE (1 << 1) +#define I2S_TX_CLOCK_ENABLE (1 << 0) + +#define I2S_TX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_TX_DESC_OFF_LEVEL_MASK (0x0F << I2S_TX_DESC_OFF_LEVEL_SHIFT) +#define I2S_TX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_TX_DESC_IFF_LEVEL_MASK (0x0F << I2S_TX_DESC_IFF_LEVEL_SHIFT) +#define I2S_TX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_TX_DESC_OFF_INTR_EN I2S_TX_DESC_OFF_INTR_EN_MSK + +#define I2S_TX_CFG (0x0000) +#define I2S_TX_IRQ_CTL (0x0004) +#define I2S_TX_IRQ_EN (0x0008) +#define I2S_TX_IRQ_IFF_THLD (0x000c) +#define I2S_TX_IRQ_OFF_THLD (0x0010) +#define I2S_TX_DESC_IFF_ADDR (0x0014) +#define I2S_TX_DESC_IFF_LEN (0x0018) +#define I2S_TX_DESC_OFF_ADDR (0x001C) +#define I2S_TX_DESC_OFF_LEN (0x0020) +#define I2S_TX_CFG_2 (0x0024) +#define I2S_TX_SLAVE_MODE_SHIFT 13 +#define I2S_TX_SLAVE_MODE_MASK (1 << I2S_TX_SLAVE_MODE_SHIFT) +#define I2S_TX_SLAVE_MODE I2S_TX_SLAVE_MODE_MASK +#define I2S_TX_MASTER_MODE 0 +#define I2S_TX_INTR_MASK 0x0F + +#define I2S_RX_ENABLE_MASK (1 << 31) +#define I2S_RX_ENABLE I2S_RX_ENABLE_MASK +#define I2S_RX_IN_R (1 << 19) +#define I2S_RX_DATA_ALIGNMENT (1 << 2) +#define I2S_RX_CLOCK_ENABLE (1 << 0) + +#define I2S_RX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_RX_DESC_OFF_LEVEL_MASK (0x0F << I2S_RX_DESC_OFF_LEVEL_SHIFT) +#define I2S_RX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_RX_DESC_IFF_LEVEL_MASK (0x0F << I2S_RX_DESC_IFF_LEVEL_SHIFT) +#define I2S_RX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_RX_DESC_OFF_INTR_EN I2S_RX_DESC_OFF_INTR_EN_MSK + +#define I2S_RX_CFG (0x0040) /* 20c0 */ +#define I2S_RX_IRQ_CTL (0x0044) +#define I2S_RX_IRQ_EN (0x0048) +#define I2S_RX_IRQ_IFF_THLD (0x004C) +#define I2S_RX_IRQ_OFF_THLD (0x0050) +#define I2S_RX_DESC_IFF_ADDR (0x0054) +#define I2S_RX_DESC_IFF_LEN (0x0058) +#define I2S_RX_DESC_OFF_ADDR (0x005C) +#define I2S_RX_DESC_OFF_LEN (0x0060) +#define I2S_RX_CFG_2 (0x0064) +#define I2S_RX_SLAVE_MODE_SHIFT 13 +#define I2S_RX_SLAVE_MODE_MASK (1 << I2S_RX_SLAVE_MODE_SHIFT) +#define I2S_RX_SLAVE_MODE I2S_RX_SLAVE_MODE_MASK +#define I2S_RX_MASTER_MODE 0 +#define I2S_RX_INTR_MASK 0x0F + +#define I2S_REG_MAX 0x007C + +struct bcm_i2s_priv { + struct device *dev; + struct resource *r_irq; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + struct snd_pcm_substream *play_substream; + struct snd_pcm_substream *capture_substream; + struct i2s_dma_desc *play_dma_desc; + struct i2s_dma_desc *capture_dma_desc; +}; + +extern int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv); +extern int bcm63xx_soc_platform_remove(struct platform_device *pdev); + +#endif diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c new file mode 100644 index 000000000000..e46c390683e7 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -0,0 +1,485 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// linux/sound/bcm/bcm63xx-pcm-whistler.c +// BCM63xx whistler pcm interface +// Copyright (c) 2020 Broadcom Corporation +// Author: Kevin-Ke Li <kevin-ke.li@broadcom.com> + +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include "bcm63xx-i2s.h" + + +struct i2s_dma_desc { + unsigned char *dma_area; + dma_addr_t dma_addr; + unsigned int dma_len; +}; + +struct bcm63xx_runtime_data { + int dma_len; + dma_addr_t dma_addr; + dma_addr_t dma_addr_next; +}; + +static const struct snd_pcm_hardware bcm63xx_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S32_LE, /* support S32 only */ + .period_bytes_max = 8192 - 32, + .periods_min = 1, + .periods_max = PAGE_SIZE/sizeof(struct i2s_dma_desc), + .buffer_bytes_max = 128 * 1024, + .fifo_size = 32, +}; + +static int bcm63xx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + dma_desc = kzalloc(sizeof(*dma_desc), GFP_NOWAIT); + if (!dma_desc) + return -ENOMEM; + + snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_desc); + + return 0; +} + +static int bcm63xx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + kfree(dma_desc); + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int bcm63xx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + + rtd = substream->private_data; + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_TX_IRQ_EN, + I2S_TX_DESC_OFF_INTR_EN, + I2S_TX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + I2S_TX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_write(regmap_i2s, + I2S_TX_IRQ_EN, + 0); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } else { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + I2S_RX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + I2S_RX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + 0); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } + return ret; +} + +static int bcm63xx_pcm_prepare(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct regmap *regmap_i2s; + struct bcm_i2s_priv *i2s_priv; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + uint32_t regaddr_desclen, regaddr_descaddr; + + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + dma_desc->dma_len = snd_pcm_lib_period_bytes(substream); + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regaddr_desclen = I2S_TX_DESC_IFF_LEN; + regaddr_descaddr = I2S_TX_DESC_IFF_ADDR; + } else { + regaddr_desclen = I2S_RX_DESC_IFF_LEN; + regaddr_descaddr = I2S_RX_DESC_IFF_ADDR; + } + + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len); + regmap_write(regmap_i2s, regaddr_descaddr, dma_desc->dma_addr); + + return 0; +} + +static snd_pcm_uframes_t +bcm63xx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + snd_pcm_uframes_t x; + struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; + + if ((void *)prtd->dma_addr_next == NULL) + prtd->dma_addr_next = substream->runtime->dma_addr; + + x = bytes_to_frames(substream->runtime, + prtd->dma_addr_next - substream->runtime->dma_addr); + + return x == substream->runtime->buffer_size ? 0 : x; +} + +static int bcm63xx_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_wc(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + +} + +static int bcm63xx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd; + + runtime->hw = bcm63xx_pcm_hardware; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + ret = -ENOMEM; + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + goto out; + + runtime->private_data = prtd; + return 0; +out: + return ret; +} + +static int bcm63xx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + return 0; +} + +static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) +{ + unsigned int availdepth, ifflevel, offlevel, int_status, val_1, val_2; + struct bcm63xx_runtime_data *prtd; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct regmap *regmap_i2s; + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + + i2s_priv = (struct bcm_i2s_priv *)bcm_i2s_priv; + regmap_i2s = i2s_priv->regmap_i2s; + + /* rx */ + regmap_read(regmap_i2s, I2S_RX_IRQ_CTL, &int_status); + + if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->capture_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + + offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> + I2S_RX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2); + offlevel--; + } + prtd->dma_addr_next = val_1 + val_2; + ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >> + I2S_RX_DESC_IFF_LEVEL_SHIFT; + + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_RX_IRQ_CTL, + I2S_RX_INTR_MASK, 0); + } + + /* tx */ + regmap_read(regmap_i2s, I2S_TX_IRQ_CTL, &int_status); + + if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->play_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); + + offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >> + I2S_TX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_LEN, &val_2); + prtd->dma_addr_next = val_1 + val_2; + offlevel--; + } + + ifflevel = (int_status & I2S_TX_DESC_IFF_LEVEL_MASK) >> + I2S_TX_DESC_IFF_LEVEL_SHIFT; + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_TX_IRQ_CTL, + I2S_TX_INTR_MASK, 0); + } + + return IRQ_HANDLED; +} + +static int bcm63xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bcm63xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_wc(pcm->card->dev, + size, &buf->addr, + GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static int bcm63xx_soc_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct bcm_i2s_priv *i2s_priv; + int ret; + + i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); + + of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1); + + ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + + i2s_priv->play_substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + i2s_priv->capture_substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + } + +out: + return ret; +} + +static void bcm63xx_pcm_free_dma_buffers(struct snd_soc_component *component, + struct snd_pcm *pcm) +{ + int stream; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_wc(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static const struct snd_soc_component_driver bcm63xx_soc_platform = { + .open = bcm63xx_pcm_open, + .close = bcm63xx_pcm_close, + .hw_params = bcm63xx_pcm_hw_params, + .hw_free = bcm63xx_pcm_hw_free, + .prepare = bcm63xx_pcm_prepare, + .trigger = bcm63xx_pcm_trigger, + .pointer = bcm63xx_pcm_pointer, + .mmap = bcm63xx_pcm_mmap, + .pcm_construct = bcm63xx_soc_pcm_new, + .pcm_destruct = bcm63xx_pcm_free_dma_buffers, +}; + +int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv) +{ + int ret; + + i2s_priv->r_irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!i2s_priv->r_irq) { + dev_err(&pdev->dev, "Unable to get register irq resource.\n"); + return -ENODEV; + } + + ret = devm_request_irq(&pdev->dev, i2s_priv->r_irq->start, i2s_dma_isr, + i2s_priv->r_irq->flags, "i2s_dma", (void *)i2s_priv); + if (ret) { + dev_err(&pdev->dev, + "i2s_init: failed to request interrupt.ret=%d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_soc_platform, NULL, 0); +} + +int bcm63xx_soc_platform_remove(struct platform_device *pdev) +{ + return 0; +} + +MODULE_AUTHOR("Kevin,Li <kevin-ke.li@broadcom.com>"); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC PCM Interface"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 3a80c613bc3f..f96d27c8b301 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -209,7 +209,7 @@ static struct cygnus_aio_port *cygnus_dai_get_dma_data( { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream); + return snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(soc_runtime, 0), substream); } static void ringbuf_set_initial(void __iomem *audio_io, @@ -359,7 +359,7 @@ static void disable_intr(struct snd_pcm_substream *substream) aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s on port %d\n", __func__, aio->portnum); /* The port number maps to the bit position to be set */ set_mask = BIT(aio->portnum); @@ -590,7 +590,7 @@ static int cygnus_pcm_open(struct snd_soc_component *component, if (!aio) return -ENODEV; - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw); @@ -623,7 +623,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component, aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) aio->play_stream = NULL; @@ -631,7 +631,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component, aio->capture_stream = NULL; if (!aio->play_stream && !aio->capture_stream) - dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "freed port %d\n", aio->portnum); return 0; } @@ -645,7 +645,7 @@ static int cygnus_pcm_hw_params(struct snd_soc_component *component, struct cygnus_aio_port *aio; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); @@ -660,7 +660,7 @@ static int cygnus_pcm_hw_free(struct snd_soc_component *component, struct cygnus_aio_port *aio; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -678,12 +678,12 @@ static int cygnus_pcm_prepare(struct snd_soc_component *component, struct ringbuf_regs *p_rbuf = NULL; aio = cygnus_dai_get_dma_data(substream); - dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum); bufsize = snd_pcm_lib_buffer_bytes(substream); periodsize = snd_pcm_lib_period_bytes(substream); - dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n", + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s (buf_size %lu) (period_size %lu)\n", __func__, bufsize, periodsize); configure_ringbuf_regs(substream); @@ -745,11 +745,11 @@ static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); - dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%zx @ %pK\n", + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: size 0x%zx @ %pK\n", __func__, size, buf->area); if (!buf->area) { - dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "%s: dma_alloc failed\n", __func__); return -ENOMEM; } buf->bytes = size; diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 10961190068e..ccf65f087ea6 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -23,8 +23,8 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; unsigned int mclk_rate; unsigned int rate = params_rate(params); diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 70c2f3e08d6d..cb133e80b7c3 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -23,8 +23,8 @@ static int snappercl15_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int err; err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea912439e446..e6a0c5d05fa5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -14,262 +14,264 @@ menu "CODEC drivers" config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST - select SND_SOC_88PM860X if MFD_88PM860X - select SND_SOC_L3 - select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC - select SND_SOC_AD1836 if SPI_MASTER - select SND_SOC_AD193X_SPI if SPI_MASTER - select SND_SOC_AD193X_I2C if I2C - select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_AD73311 - select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAU1761_I2C if I2C - select SND_SOC_ADAU1761_SPI if SPI - select SND_SOC_ADAU1781_I2C if I2C - select SND_SOC_ADAU1781_SPI if SPI - select SND_SOC_ADAV801 if SPI_MASTER - select SND_SOC_ADAV803 if I2C - select SND_SOC_ADAU1977_SPI if SPI_MASTER - select SND_SOC_ADAU1977_I2C if I2C - select SND_SOC_ADAU1701 if I2C - select SND_SOC_ADAU7002 - select SND_SOC_ADAU7118_I2C if I2C - select SND_SOC_ADAU7118_HW - select SND_SOC_ADS117X - select SND_SOC_AK4104 if SPI_MASTER - select SND_SOC_AK4118 if I2C - select SND_SOC_AK4458 if I2C - select SND_SOC_AK4535 if I2C - select SND_SOC_AK4554 - select SND_SOC_AK4613 if I2C - select SND_SOC_AK4641 if I2C - select SND_SOC_AK4642 if I2C - select SND_SOC_AK4671 if I2C - select SND_SOC_AK5386 - select SND_SOC_AK5558 if I2C - select SND_SOC_ALC5623 if I2C - select SND_SOC_ALC5632 if I2C - select SND_SOC_BT_SCO - select SND_SOC_BD28623 - select SND_SOC_CQ0093VC - select SND_SOC_CROS_EC_CODEC if CROS_EC - select SND_SOC_CS35L32 if I2C - select SND_SOC_CS35L33 if I2C - select SND_SOC_CS35L34 if I2C - select SND_SOC_CS35L35 if I2C - select SND_SOC_CS35L36 if I2C - select SND_SOC_CS42L42 if I2C - select SND_SOC_CS42L51_I2C if I2C - select SND_SOC_CS42L52 if I2C && INPUT - select SND_SOC_CS42L56 if I2C && INPUT - select SND_SOC_CS42L73 if I2C - select SND_SOC_CS4265 if I2C - select SND_SOC_CS4270 if I2C - select SND_SOC_CS4271_I2C if I2C - select SND_SOC_CS4271_SPI if SPI_MASTER - select SND_SOC_CS42XX8_I2C if I2C - select SND_SOC_CS43130 if I2C - select SND_SOC_CS4341 if SND_SOC_I2C_AND_SPI - select SND_SOC_CS4349 if I2C - select SND_SOC_CS47L15 if MFD_CS47L15 - select SND_SOC_CS47L24 if MFD_CS47L24 - select SND_SOC_CS47L35 if MFD_CS47L35 - select SND_SOC_CS47L85 if MFD_CS47L85 - select SND_SOC_CS47L90 if MFD_CS47L90 - select SND_SOC_CS47L92 if MFD_CS47L92 - select SND_SOC_CS53L30 if I2C - select SND_SOC_CX20442 if TTY - select SND_SOC_CX2072X if I2C - select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI - select SND_SOC_DA7213 if I2C - select SND_SOC_DA7218 if I2C - select SND_SOC_DA7219 if I2C - select SND_SOC_DA732X if I2C - select SND_SOC_DA9055 if I2C - select SND_SOC_DMIC if GPIOLIB - select SND_SOC_ES8316 if I2C - select SND_SOC_ES8328_SPI if SPI_MASTER - select SND_SOC_ES8328_I2C if I2C - select SND_SOC_ES7134 - select SND_SOC_ES7241 - select SND_SOC_GTM601 - select SND_SOC_HDAC_HDMI - select SND_SOC_HDAC_HDA - select SND_SOC_ICS43432 - select SND_SOC_INNO_RK3036 - select SND_SOC_ISABELLE if I2C - select SND_SOC_JZ4740_CODEC - select SND_SOC_JZ4725B_CODEC - select SND_SOC_JZ4770_CODEC - select SND_SOC_LM4857 if I2C - select SND_SOC_LM49453 if I2C - select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR - select SND_SOC_MAX98088 if I2C - select SND_SOC_MAX98090 if I2C - select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A if GPIOLIB - select SND_SOC_MAX98371 if I2C - select SND_SOC_MAX98504 if I2C - select SND_SOC_MAX9867 if I2C - select SND_SOC_MAX98925 if I2C - select SND_SOC_MAX98926 if I2C - select SND_SOC_MAX98927 if I2C - select SND_SOC_MAX98373 if I2C - select SND_SOC_MAX9850 if I2C - select SND_SOC_MAX9860 if I2C - select SND_SOC_MAX9759 - select SND_SOC_MAX9768 if I2C - select SND_SOC_MAX9877 if I2C - select SND_SOC_MC13783 if MFD_MC13XXX - select SND_SOC_ML26124 if I2C - select SND_SOC_MT6351 if MTK_PMIC_WRAP - select SND_SOC_MT6358 if MTK_PMIC_WRAP - select SND_SOC_MT6660 if I2C - select SND_SOC_NAU8540 if I2C - select SND_SOC_NAU8810 if I2C - select SND_SOC_NAU8822 if I2C - select SND_SOC_NAU8824 if I2C - select SND_SOC_NAU8825 if I2C - select SND_SOC_HDMI_CODEC - select SND_SOC_PCM1681 if I2C - select SND_SOC_PCM1789_I2C if I2C - select SND_SOC_PCM179X_I2C if I2C - select SND_SOC_PCM179X_SPI if SPI_MASTER - select SND_SOC_PCM186X_I2C if I2C - select SND_SOC_PCM186X_SPI if SPI_MASTER - select SND_SOC_PCM3008 - select SND_SOC_PCM3060_I2C if I2C - select SND_SOC_PCM3060_SPI if SPI_MASTER - select SND_SOC_PCM3168A_I2C if I2C - select SND_SOC_PCM3168A_SPI if SPI_MASTER - select SND_SOC_PCM5102A - select SND_SOC_PCM512x_I2C if I2C - select SND_SOC_PCM512x_SPI if SPI_MASTER - select SND_SOC_RK3328 - select SND_SOC_RT274 if I2C - select SND_SOC_RT286 if I2C - select SND_SOC_RT298 if I2C - select SND_SOC_RT1011 if I2C - select SND_SOC_RT1015 if I2C - select SND_SOC_RT1305 if I2C - select SND_SOC_RT1308 if I2C - select SND_SOC_RT5514 if I2C - select SND_SOC_RT5616 if I2C - select SND_SOC_RT5631 if I2C - select SND_SOC_RT5640 if I2C - select SND_SOC_RT5645 if I2C - select SND_SOC_RT5651 if I2C - select SND_SOC_RT5659 if I2C - select SND_SOC_RT5660 if I2C - select SND_SOC_RT5663 if I2C - select SND_SOC_RT5665 if I2C - select SND_SOC_RT5668 if I2C - select SND_SOC_RT5670 if I2C - select SND_SOC_RT5677 if I2C && SPI_MASTER - select SND_SOC_RT5682 if I2C - select SND_SOC_RT700_SDW if SOUNDWIRE - select SND_SOC_RT711_SDW if SOUNDWIRE - select SND_SOC_RT715_SDW if SOUNDWIRE - select SND_SOC_RT1308_SDW if SOUNDWIRE - select SND_SOC_SGTL5000 if I2C - select SND_SOC_SI476X if MFD_SI476X_CORE - select SND_SOC_SIMPLE_AMPLIFIER - select SND_SOC_SIRF_AUDIO_CODEC - select SND_SOC_SPDIF - select SND_SOC_SSM2305 - select SND_SOC_SSM2518 if I2C - select SND_SOC_SSM2602_SPI if SPI_MASTER - select SND_SOC_SSM2602_I2C if I2C - select SND_SOC_SSM4567 if I2C - select SND_SOC_STA32X if I2C - select SND_SOC_STA350 if I2C - select SND_SOC_STA529 if I2C - select SND_SOC_STAC9766 if SND_SOC_AC97_BUS - select SND_SOC_STI_SAS - select SND_SOC_TAS2552 if I2C - select SND_SOC_TAS2562 if I2C - select SND_SOC_TAS2770 if I2C - select SND_SOC_TAS5086 if I2C - select SND_SOC_TAS571X if I2C - select SND_SOC_TAS5720 if I2C - select SND_SOC_TAS6424 if I2C - select SND_SOC_TDA7419 if I2C - select SND_SOC_TFA9879 if I2C - select SND_SOC_TLV320AIC23_I2C if I2C - select SND_SOC_TLV320AIC23_SPI if SPI_MASTER - select SND_SOC_TLV320AIC26 if SPI_MASTER - select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK - select SND_SOC_TLV320AIC3X if I2C - select SND_SOC_TPA6130A2 if I2C - select SND_SOC_TLV320DAC33 if I2C - select SND_SOC_TSCS42XX if I2C - select SND_SOC_TSCS454 if I2C - select SND_SOC_TS3A227E if I2C - select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL6040_CORE - select SND_SOC_UDA1334 if GPIOLIB - select SND_SOC_UDA134X - select SND_SOC_UDA1380 if I2C - select SND_SOC_WCD9335 if SLIMBUS - select SND_SOC_WCD934X if MFD_WCD934X && COMMON_CLK - select SND_SOC_WL1273 if MFD_WL1273_CORE - select SND_SOC_WM0010 if SPI_MASTER - select SND_SOC_WM1250_EV1 if I2C - select SND_SOC_WM2000 if I2C - select SND_SOC_WM2200 if I2C - select SND_SOC_WM5100 if I2C - select SND_SOC_WM5102 if MFD_WM5102 - select SND_SOC_WM5110 if MFD_WM5110 - select SND_SOC_WM8350 if MFD_WM8350 - select SND_SOC_WM8400 if MFD_WM8400 - select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8523 if I2C - select SND_SOC_WM8524 if GPIOLIB - select SND_SOC_WM8580 if I2C - select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8727 - select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8737 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8770 if SPI_MASTER - select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8782 - select SND_SOC_WM8804_I2C if I2C - select SND_SOC_WM8804_SPI if SPI_MASTER - select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C - select SND_SOC_WM8904 if I2C - select SND_SOC_WM8940 if I2C - select SND_SOC_WM8955 if I2C - select SND_SOC_WM8960 if I2C - select SND_SOC_WM8961 if I2C - select SND_SOC_WM8962 if I2C && INPUT - select SND_SOC_WM8971 if I2C - select SND_SOC_WM8974 if I2C - select SND_SOC_WM8978 if I2C - select SND_SOC_WM8983 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8990 if I2C - select SND_SOC_WM8991 if I2C - select SND_SOC_WM8993 if I2C - select SND_SOC_WM8994 if MFD_WM8994 - select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI - select SND_SOC_WM8996 if I2C - select SND_SOC_WM8997 if MFD_WM8997 - select SND_SOC_WM8998 if MFD_WM8998 - select SND_SOC_WM9081 if I2C - select SND_SOC_WM9090 if I2C - select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - select SND_SOC_WSA881X if SOUNDWIRE + imply SND_SOC_88PM860X + imply SND_SOC_L3 + imply SND_SOC_AB8500_CODEC + imply SND_SOC_AC97_CODEC + imply SND_SOC_AD1836 + imply SND_SOC_AD193X_SPI + imply SND_SOC_AD193X_I2C + imply SND_SOC_AD1980 + imply SND_SOC_AD73311 + imply SND_SOC_ADAU1373 + imply SND_SOC_ADAU1761_I2C + imply SND_SOC_ADAU1761_SPI + imply SND_SOC_ADAU1781_I2C + imply SND_SOC_ADAU1781_SPI + imply SND_SOC_ADAV801 + imply SND_SOC_ADAV803 + imply SND_SOC_ADAU1977_SPI + imply SND_SOC_ADAU1977_I2C + imply SND_SOC_ADAU1701 + imply SND_SOC_ADAU7002 + imply SND_SOC_ADAU7118_I2C + imply SND_SOC_ADAU7118_HW + imply SND_SOC_ADS117X + imply SND_SOC_AK4104 + imply SND_SOC_AK4118 + imply SND_SOC_AK4458 + imply SND_SOC_AK4535 + imply SND_SOC_AK4554 + imply SND_SOC_AK4613 + imply SND_SOC_AK4641 + imply SND_SOC_AK4642 + imply SND_SOC_AK4671 + imply SND_SOC_AK5386 + imply SND_SOC_AK5558 + imply SND_SOC_ALC5623 + imply SND_SOC_ALC5632 + imply SND_SOC_BT_SCO + imply SND_SOC_BD28623 + imply SND_SOC_CQ0093VC + imply SND_SOC_CROS_EC_CODEC + imply SND_SOC_CS35L32 + imply SND_SOC_CS35L33 + imply SND_SOC_CS35L34 + imply SND_SOC_CS35L35 + imply SND_SOC_CS35L36 + imply SND_SOC_CS42L42 + imply SND_SOC_CS42L51_I2C + imply SND_SOC_CS42L52 + imply SND_SOC_CS42L56 + imply SND_SOC_CS42L73 + imply SND_SOC_CS4265 + imply SND_SOC_CS4270 + imply SND_SOC_CS4271_I2C + imply SND_SOC_CS4271_SPI + imply SND_SOC_CS42XX8_I2C + imply SND_SOC_CS43130 + imply SND_SOC_CS4341 + imply SND_SOC_CS4349 + imply SND_SOC_CS47L15 + imply SND_SOC_CS47L24 + imply SND_SOC_CS47L35 + imply SND_SOC_CS47L85 + imply SND_SOC_CS47L90 + imply SND_SOC_CS47L92 + imply SND_SOC_CS53L30 + imply SND_SOC_CX20442 + imply SND_SOC_CX2072X + imply SND_SOC_DA7210 + imply SND_SOC_DA7213 + imply SND_SOC_DA7218 + imply SND_SOC_DA7219 + imply SND_SOC_DA732X + imply SND_SOC_DA9055 + imply SND_SOC_DMIC + imply SND_SOC_ES8316 + imply SND_SOC_ES8328_SPI + imply SND_SOC_ES8328_I2C + imply SND_SOC_ES7134 + imply SND_SOC_ES7241 + imply SND_SOC_GTM601 + imply SND_SOC_HDAC_HDMI + imply SND_SOC_HDAC_HDA + imply SND_SOC_ICS43432 + imply SND_SOC_INNO_RK3036 + imply SND_SOC_ISABELLE + imply SND_SOC_JZ4740_CODEC + imply SND_SOC_JZ4725B_CODEC + imply SND_SOC_JZ4770_CODEC + imply SND_SOC_LM4857 + imply SND_SOC_LM49453 + imply SND_SOC_LOCHNAGAR_SC + imply SND_SOC_MAX98088 + imply SND_SOC_MAX98090 + imply SND_SOC_MAX98095 + imply SND_SOC_MAX98357A + imply SND_SOC_MAX98371 + imply SND_SOC_MAX98504 + imply SND_SOC_MAX9867 + imply SND_SOC_MAX98925 + imply SND_SOC_MAX98926 + imply SND_SOC_MAX98927 + imply SND_SOC_MAX98373 + imply SND_SOC_MAX9850 + imply SND_SOC_MAX9860 + imply SND_SOC_MAX9759 + imply SND_SOC_MAX9768 + imply SND_SOC_MAX9877 + imply SND_SOC_MC13783 + imply SND_SOC_ML26124 + imply SND_SOC_MT6351 + imply SND_SOC_MT6358 + imply SND_SOC_MT6660 + imply SND_SOC_NAU8540 + imply SND_SOC_NAU8810 + imply SND_SOC_NAU8822 + imply SND_SOC_NAU8824 + imply SND_SOC_NAU8825 + imply SND_SOC_HDMI_CODEC + imply SND_SOC_PCM1681 + imply SND_SOC_PCM1789_I2C + imply SND_SOC_PCM179X_I2C + imply SND_SOC_PCM179X_SPI + imply SND_SOC_PCM186X_I2C + imply SND_SOC_PCM186X_SPI + imply SND_SOC_PCM3008 + imply SND_SOC_PCM3060_I2C + imply SND_SOC_PCM3060_SPI + imply SND_SOC_PCM3168A_I2C + imply SND_SOC_PCM3168A_SPI + imply SND_SOC_PCM5102A + imply SND_SOC_PCM512x_I2C + imply SND_SOC_PCM512x_SPI + imply SND_SOC_RK3328 + imply SND_SOC_RT274 + imply SND_SOC_RT286 + imply SND_SOC_RT298 + imply SND_SOC_RT1011 + imply SND_SOC_RT1015 + imply SND_SOC_RT1305 + imply SND_SOC_RT1308 + imply SND_SOC_RT5514 + imply SND_SOC_RT5616 + imply SND_SOC_RT5631 + imply SND_SOC_RT5640 + imply SND_SOC_RT5645 + imply SND_SOC_RT5651 + imply SND_SOC_RT5659 + imply SND_SOC_RT5660 + imply SND_SOC_RT5663 + imply SND_SOC_RT5665 + imply SND_SOC_RT5668 + imply SND_SOC_RT5670 + imply SND_SOC_RT5677 + imply SND_SOC_RT5682 + imply SND_SOC_RT5682_SDW + imply SND_SOC_RT700_SDW + imply SND_SOC_RT711_SDW + imply SND_SOC_RT715_SDW + imply SND_SOC_RT1308_SDW + imply SND_SOC_SGTL5000 + imply SND_SOC_SI476X + imply SND_SOC_SIMPLE_AMPLIFIER + imply SND_SOC_SIRF_AUDIO_CODEC + imply SND_SOC_SPDIF + imply SND_SOC_SSM2305 + imply SND_SOC_SSM2518 + imply SND_SOC_SSM2602_SPI + imply SND_SOC_SSM2602_I2C + imply SND_SOC_SSM4567 + imply SND_SOC_STA32X + imply SND_SOC_STA350 + imply SND_SOC_STA529 + imply SND_SOC_STAC9766 + imply SND_SOC_STI_SAS + imply SND_SOC_TAS2552 + imply SND_SOC_TAS2562 + imply SND_SOC_TAS2770 + imply SND_SOC_TAS5086 + imply SND_SOC_TAS571X + imply SND_SOC_TAS5720 + imply SND_SOC_TAS6424 + imply SND_SOC_TDA7419 + imply SND_SOC_TFA9879 + imply SND_SOC_TLV320ADCX140 + imply SND_SOC_TLV320AIC23_I2C + imply SND_SOC_TLV320AIC23_SPI + imply SND_SOC_TLV320AIC26 + imply SND_SOC_TLV320AIC31XX + imply SND_SOC_TLV320AIC32X4_I2C + imply SND_SOC_TLV320AIC32X4_SPI + imply SND_SOC_TLV320AIC3X + imply SND_SOC_TPA6130A2 + imply SND_SOC_TLV320DAC33 + imply SND_SOC_TSCS42XX + imply SND_SOC_TSCS454 + imply SND_SOC_TS3A227E + imply SND_SOC_TWL4030 + imply SND_SOC_TWL6040 + imply SND_SOC_UDA1334 + imply SND_SOC_UDA134X + imply SND_SOC_UDA1380 + imply SND_SOC_WCD9335 + imply SND_SOC_WCD934X + imply SND_SOC_WL1273 + imply SND_SOC_WM0010 + imply SND_SOC_WM1250_EV1 + imply SND_SOC_WM2000 + imply SND_SOC_WM2200 + imply SND_SOC_WM5100 + imply SND_SOC_WM5102 + imply SND_SOC_WM5110 + imply SND_SOC_WM8350 + imply SND_SOC_WM8400 + imply SND_SOC_WM8510 + imply SND_SOC_WM8523 + imply SND_SOC_WM8524 + imply SND_SOC_WM8580 + imply SND_SOC_WM8711 + imply SND_SOC_WM8727 + imply SND_SOC_WM8728 + imply SND_SOC_WM8731 + imply SND_SOC_WM8737 + imply SND_SOC_WM8741 + imply SND_SOC_WM8750 + imply SND_SOC_WM8753 + imply SND_SOC_WM8770 + imply SND_SOC_WM8776 + imply SND_SOC_WM8782 + imply SND_SOC_WM8804_I2C + imply SND_SOC_WM8804_SPI + imply SND_SOC_WM8900 + imply SND_SOC_WM8903 + imply SND_SOC_WM8904 + imply SND_SOC_WM8940 + imply SND_SOC_WM8955 + imply SND_SOC_WM8960 + imply SND_SOC_WM8961 + imply SND_SOC_WM8962 + imply SND_SOC_WM8971 + imply SND_SOC_WM8974 + imply SND_SOC_WM8978 + imply SND_SOC_WM8983 + imply SND_SOC_WM8985 + imply SND_SOC_WM8988 + imply SND_SOC_WM8990 + imply SND_SOC_WM8991 + imply SND_SOC_WM8993 + imply SND_SOC_WM8994 + imply SND_SOC_WM8995 + imply SND_SOC_WM8996 + imply SND_SOC_WM8997 + imply SND_SOC_WM8998 + imply SND_SOC_WM9081 + imply SND_SOC_WM9090 + imply SND_SOC_WM9705 + imply SND_SOC_WM9712 + imply SND_SOC_WM9713 + imply SND_SOC_WSA881X help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -283,6 +285,7 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate + depends on MFD_88PM860X config SND_SOC_ARIZONA tristate @@ -318,6 +321,7 @@ config SND_SOC_WM_ADSP config SND_SOC_AB8500_CODEC tristate + depends on ABX500_CORE config SND_SOC_AC97_CODEC tristate "Build generic ASoC AC97 CODEC driver" @@ -326,21 +330,25 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1836 tristate + depends on SPI_MASTER config SND_SOC_AD193X tristate config SND_SOC_AD193X_SPI tristate + depends on SPI_MASTER select SND_SOC_AD193X config SND_SOC_AD193X_I2C tristate + depends on I2C select SND_SOC_AD193X config SND_SOC_AD1980 - select REGMAP_AC97 tristate + depends on SND_SOC_AC97_BUS + select REGMAP_AC97 config SND_SOC_AD73311 tristate @@ -350,6 +358,7 @@ config SND_SOC_ADAU_UTILS config SND_SOC_ADAU1373 tristate + depends on I2C select SND_SOC_ADAU_UTILS config SND_SOC_ADAU1701 @@ -384,11 +393,13 @@ config SND_SOC_ADAU1781 config SND_SOC_ADAU1781_I2C tristate + depends on I2C select SND_SOC_ADAU1781 select REGMAP_I2C config SND_SOC_ADAU1781_SPI tristate + depends on SPI_MASTER select SND_SOC_ADAU1781 select REGMAP_SPI @@ -397,11 +408,13 @@ config SND_SOC_ADAU1977 config SND_SOC_ADAU1977_SPI tristate + depends on SPI_MASTER select SND_SOC_ADAU1977 select REGMAP_SPI config SND_SOC_ADAU1977_I2C tristate + depends on I2C select SND_SOC_ADAU1977 select REGMAP_I2C @@ -440,10 +453,12 @@ config SND_SOC_ADAV80X config SND_SOC_ADAV801 tristate + depends on SPI_MASTER select SND_SOC_ADAV80X config SND_SOC_ADAV803 tristate + depends on I2C select SND_SOC_ADAV80X config SND_SOC_ADS117X @@ -465,6 +480,7 @@ config SND_SOC_AK4458 config SND_SOC_AK4535 tristate + depends on I2C config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" @@ -475,6 +491,7 @@ config SND_SOC_AK4613 config SND_SOC_AK4641 tristate + depends on I2C config SND_SOC_AK4642 tristate "AKM AK4642 CODEC" @@ -482,6 +499,7 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate + depends on I2C config SND_SOC_AK5386 tristate "AKM AK5638 CODEC" @@ -497,6 +515,7 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate + depends on I2C config SND_SOC_BD28623 tristate "ROHM BD28623 CODEC" @@ -631,6 +650,7 @@ config SND_SOC_CS47L15 config SND_SOC_CS47L24 tristate + depends on MFD_CS47L24 config SND_SOC_CS47L35 tristate @@ -697,6 +717,7 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate + depends on I2C config SND_SOC_DA7213 tristate "Dialog DA7213 CODEC" @@ -704,15 +725,19 @@ config SND_SOC_DA7213 config SND_SOC_DA7218 tristate + depends on I2C config SND_SOC_DA7219 tristate + depends on I2C config SND_SOC_DA732X tristate + depends on I2C config SND_SOC_DA9055 tristate + depends on I2C config SND_SOC_DMIC tristate "Generic Digital Microphone CODEC" @@ -772,9 +797,11 @@ config SND_SOC_INNO_RK3036 config SND_SOC_ISABELLE tristate + depends on I2C config SND_SOC_LM49453 tristate + depends on I2C config SND_SOC_LOCHNAGAR_SC tristate "Lochnagar Sound Card" @@ -801,17 +828,20 @@ config SND_SOC_MAX98088 depends on I2C config SND_SOC_MAX98090 - tristate + tristate + depends on I2C config SND_SOC_MAX98095 - tristate + tristate + depends on I2C config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" depends on GPIOLIB config SND_SOC_MAX98371 - tristate + tristate + depends on I2C config SND_SOC_MAX98504 tristate "Maxim MAX98504 speaker amplifier" @@ -822,10 +852,12 @@ config SND_SOC_MAX9867 depends on I2C config SND_SOC_MAX98925 - tristate + tristate + depends on I2C config SND_SOC_MAX98926 tristate + depends on I2C config SND_SOC_MAX98927 tristate "Maxim Integrated MAX98927 Speaker Amplifier" @@ -837,6 +869,7 @@ config SND_SOC_MAX98373 config SND_SOC_MAX9850 tristate + depends on I2C config SND_SOC_MAX9860 tristate "Maxim MAX9860 Mono Audio Voice Codec" @@ -1015,26 +1048,32 @@ config SND_SOC_RT298 config SND_SOC_RT1011 tristate + depends on I2C config SND_SOC_RT1015 tristate + depends on I2C config SND_SOC_RT1305 tristate + depends on I2C config SND_SOC_RT1308 tristate + depends on I2C config SND_SOC_RT1308_SDW tristate "Realtek RT1308 Codec - SDW" - depends on SOUNDWIRE + depends on I2C && SOUNDWIRE select REGMAP_SOUNDWIRE config SND_SOC_RT5514 tristate + depends on I2C config SND_SOC_RT5514_SPI tristate + depends on SPI_MASTER config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors @@ -1050,33 +1089,43 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate + depends on I2C config SND_SOC_RT5645 tristate + depends on I2C config SND_SOC_RT5651 tristate + depends on I2C config SND_SOC_RT5659 tristate + depends on I2C config SND_SOC_RT5660 tristate + depends on I2C config SND_SOC_RT5663 tristate + depends on I2C config SND_SOC_RT5665 tristate + depends on I2C config SND_SOC_RT5668 tristate + depends on I2C config SND_SOC_RT5670 tristate + depends on I2C config SND_SOC_RT5677 tristate + depends on I2C select REGMAP_I2C select REGMAP_IRQ @@ -1086,6 +1135,13 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate + depends on I2C || SOUNDWIRE + +config SND_SOC_RT5682_SDW + tristate "Realtek RT5682 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT5682 + select REGMAP_SOUNDWIRE config SND_SOC_RT700 tristate @@ -1153,6 +1209,7 @@ config SND_SOC_SSM2305 config SND_SOC_SSM2518 tristate + depends on I2C config SND_SOC_SSM2602 tristate @@ -1184,9 +1241,11 @@ config SND_SOC_STA350 config SND_SOC_STA529 tristate + depends on I2C config SND_SOC_STAC9766 tristate + depends on SND_SOC_AC97_BUS config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" @@ -1281,6 +1340,15 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate + depends on I2C + +config SND_SOC_TLV320ADCX140 + tristate "Texas Instruments TLV320ADCX140 CODEC family" + depends on I2C + select REGMAP_I2C + help + Add support for Texas Instruments tlv320adc3140, tlv320adc5140 and + tlv320adc6140 quad channel ADCs. config SND_SOC_TS3A227E tristate "TI Headset/Mic detect and keypress chip" @@ -1301,11 +1369,13 @@ config SND_SOC_TSCS454 Add support for Tempo Semiconductor's TSCS454 audio CODEC. config SND_SOC_TWL4030 - select MFD_TWL4030_AUDIO tristate + depends on TWL4030_CORE + select MFD_TWL4030_AUDIO config SND_SOC_TWL6040 tristate + depends on TWL6040_CORE config SND_SOC_UDA1334 tristate "NXP UDA1334 DAC" @@ -1345,30 +1415,40 @@ config SND_SOC_WL1273 config SND_SOC_WM0010 tristate + depends on SPI_MASTER config SND_SOC_WM1250_EV1 tristate + depends on I2C config SND_SOC_WM2000 tristate + depends on I2C config SND_SOC_WM2200 tristate + depends on I2C config SND_SOC_WM5100 tristate + depends on I2C config SND_SOC_WM5102 tristate + depends on MFD_WM5102 config SND_SOC_WM5110 tristate + depends on MFD_WM5110 config SND_SOC_WM8350 tristate + depends on MFD_WM8350 config SND_SOC_WM8400 tristate + # FIXME nothing selects SND_SOC_WM8400?? + depends on MFD_WM8400 config SND_SOC_WM8510 tristate "Wolfson Microelectronics WM8510 CODEC" @@ -1456,9 +1536,11 @@ config SND_SOC_WM8904 config SND_SOC_WM8940 tristate + depends on I2C config SND_SOC_WM8955 tristate + depends on I2C config SND_SOC_WM8960 tristate "Wolfson Microelectronics WM8960 CODEC" @@ -1466,6 +1548,7 @@ config SND_SOC_WM8960 config SND_SOC_WM8961 tristate + depends on I2C config SND_SOC_WM8962 tristate "Wolfson Microelectronics WM8962 CODEC" @@ -1473,6 +1556,7 @@ config SND_SOC_WM8962 config SND_SOC_WM8971 tristate + depends on I2C config SND_SOC_WM8974 tristate "Wolfson Microelectronics WM8974 codec" @@ -1484,6 +1568,7 @@ config SND_SOC_WM8978 config SND_SOC_WM8983 tristate + depends on I2C config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" @@ -1494,12 +1579,15 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate + depends on I2C config SND_SOC_WM8991 tristate + depends on I2C config SND_SOC_WM8993 tristate + depends on I2C config SND_SOC_WM8994 tristate @@ -1509,12 +1597,15 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate + depends on I2C config SND_SOC_WM8997 tristate + depends on MFD_WM8997 config SND_SOC_WM8998 tristate + depends on MFD_WM8998 config SND_SOC_WM9081 tristate @@ -1522,19 +1613,23 @@ config SND_SOC_WM9081 config SND_SOC_WM9090 tristate + depends on I2C config SND_SOC_WM9705 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW @@ -1555,6 +1650,7 @@ config SND_SOC_ZX_AUD96P22 # Amp config SND_SOC_LM4857 tristate + depends on I2C config SND_SOC_MAX9759 tristate "Maxim MAX9759 speaker Amplifier" @@ -1562,15 +1658,19 @@ config SND_SOC_MAX9759 config SND_SOC_MAX9768 tristate + depends on I2C config SND_SOC_MAX9877 tristate + depends on I2C config SND_SOC_MC13783 tristate + depends on MFD_MC13XXX config SND_SOC_ML26124 tristate + depends on I2C config SND_SOC_MT6351 tristate "MediaTek MT6351 Codec" @@ -1608,6 +1708,7 @@ config SND_SOC_NAU8824 config SND_SOC_NAU8825 tristate + depends on I2C config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ba1b4b3fa2da..03533157cda6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -177,6 +177,7 @@ snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o +snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o @@ -218,6 +219,7 @@ snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-tlv320adcx140-objs := tlv320adcx140.o snd-soc-tscs42xx-objs := tscs42xx.o snd-soc-tscs454-objs := tscs454.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -476,6 +478,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o +obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o @@ -516,6 +519,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TLV320ADCX140) += snd-soc-tlv320adcx140.o obj-$(CONFIG_SND_SOC_TSCS42XX) += snd-soc-tscs42xx.o obj-$(CONFIG_SND_SOC_TSCS454) += snd-soc-tscs454.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 6a24f570c5e8..d3dc42aa6825 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -45,6 +45,9 @@ struct cros_ec_codec_priv { /* DMIC */ atomic_t dmic_probed; + /* I2S_RX */ + uint32_t i2s_rx_bclk_ratio; + /* WoV */ bool wov_enabled; uint8_t *wov_audio_shm_p; @@ -259,6 +262,7 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, snd_soc_component_get_drvdata(component); struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_sample_depth depth; + uint32_t bclk; int ret; if (params_rate(params) != 48000) @@ -284,15 +288,29 @@ static int i2s_rx_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - dev_dbg(component->dev, "set bclk to %u\n", - snd_soc_params_to_bclk(params)); + if (priv->i2s_rx_bclk_ratio) + bclk = params_rate(params) * priv->i2s_rx_bclk_ratio; + else + bclk = snd_soc_params_to_bclk(params); + + dev_dbg(component->dev, "set bclk to %u\n", bclk); p.cmd = EC_CODEC_I2S_RX_SET_BCLK; - p.set_bclk_param.bclk = snd_soc_params_to_bclk(params); + p.set_bclk_param.bclk = bclk; return send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_I2S_RX, (uint8_t *)&p, sizeof(p), NULL, 0); } +static int i2s_rx_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct cros_ec_codec_priv *priv = + snd_soc_component_get_drvdata(component); + + priv->i2s_rx_bclk_ratio = ratio; + return 0; +} + static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; @@ -340,6 +358,7 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) static const struct snd_soc_dai_ops i2s_rx_dai_ops = { .hw_params = i2s_rx_hw_params, .set_fmt = i2s_rx_set_fmt, + .set_bclk_ratio = i2s_rx_set_bclk_ratio, }; static int i2s_rx_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 04b86a51e055..62f412d6f9f2 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - !dai->capture_active) || + !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && - !dai->playback_active)) { + !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN, CS4271_MODE2_PDN); diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index e8840dc142ef..8d1869bf7f9c 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1239,12 +1239,12 @@ static int cs47l15_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l15-dsp-trace") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l15-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 25bffc2968f0..6b0570f59630 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1076,14 +1076,14 @@ static int cs47l24_open(struct snd_compr_stream *stream) struct arizona *arizona = priv->core.arizona; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l24-dsp-trace") == 0) { n_adsp = 1; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 3d48a0d9ecc5..18839807c9d1 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1514,14 +1514,14 @@ static int cs47l35_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "cs47l35-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l35-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index bef3471f482d..a575113207f0 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2457,14 +2457,14 @@ static int cs47l85_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-voicectrl") == 0) { n_adsp = 5; - } else if (strcmp(rtd->codec_dai->name, "cs47l85-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l85-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 266eade82764..81a1311b14e6 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2368,14 +2368,14 @@ static int cs47l90_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-voicectrl") == 0) { n_adsp = 5; - } else if (strcmp(rtd->codec_dai->name, "cs47l90-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l90-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 942040fd354f..15fc213d178d 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -1840,12 +1840,12 @@ static int cs47l92_open(struct snd_compr_stream *stream) struct madera *madera = priv->madera; int n_adsp; - if (strcmp(rtd->codec_dai->name, "cs47l92-dsp-trace") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "cs47l92-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(madera->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e6558475e006..fba9b749839d 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1998,11 +1998,11 @@ static struct hdac_hdmi_drv_data intel_drv_data = { static int hdac_hdmi_dev_probe(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi_priv = NULL; + struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int num_dais = 0; - int ret = 0; + int ret; struct hdac_driver *hdrv = drv_to_hdac_driver(hdev->dev.driver); const struct hda_device_id *hdac_id = hdac_get_device_id(hdev, hdrv); diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 16313b973eaa..a8bd793a7867 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -5,6 +5,7 @@ */ #include <linux/acpi.h> +#include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> #include <linux/gpio.h> @@ -24,26 +25,24 @@ struct max98357a_priv { unsigned int sdmode_delay; }; -static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct max98357a_priv *max98357a = + snd_soc_component_get_drvdata(component); if (!max98357a->sdmode) return 0; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - mdelay(max98357a->sdmode_delay); + if (event & SND_SOC_DAPM_POST_PMU) { + msleep(max98357a->sdmode_delay); gpiod_set_value(max98357a->sdmode, 1); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev_dbg(component->dev, "set sdmode to 1"); + } else if (event & SND_SOC_DAPM_PRE_PMD) { gpiod_set_value(max98357a->sdmode, 0); - break; + dev_dbg(component->dev, "set sdmode to 0"); } return 0; @@ -51,10 +50,14 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0, + max98357a_sdmode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { - {"Speaker", NULL, "HiFi Playback"}, + {"SD_MODE", NULL, "HiFi Playback"}, + {"Speaker", NULL, "SD_MODE"}, }; static const struct snd_soc_component_driver max98357a_component_driver = { @@ -68,10 +71,6 @@ static const struct snd_soc_component_driver max98357a_component_driver = { .non_legacy_dai_naming = 1, }; -static const struct snd_soc_dai_ops max98357a_dai_ops = { - .trigger = max98357a_daiops_trigger, -}; - static struct snd_soc_dai_driver max98357a_dai_driver = { .name = "HiFi", .playback = { @@ -91,7 +90,6 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { .channels_min = 1, .channels_max = 2, }, - .ops = &max98357a_dai_ops, }; static int max98357a_platform_probe(struct platform_device *pdev) @@ -135,6 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id max98357a_acpi_match[] = { { "MX98357A", 0 }, + { "MX98360A", 0 }, {}, }; MODULE_DEVICE_TABLE(acpi, max98357a_acpi_match); diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index a36c416caad4..d1797003c83d 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -1,15 +1,13 @@ -// SPDX-License-Identifier: GPL-2.0 // +// SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2019 MediaTek Inc. #include <linux/module.h> #include <linux/kernel.h> -#include <linux/version.h> #include <linux/err.h> #include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/delay.h> -#include <linux/debugfs.h> #include <sound/soc.h> #include <sound/tlv.h> #include <sound/pcm_params.h> @@ -225,14 +223,87 @@ static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) 0x01, on_off ? 0x00 : 0x01); } +struct reg_table { + uint32_t addr; + uint32_t mask; + uint32_t val; +}; + +static const struct reg_table mt6660_setting_table[] = { + { 0x20, 0x80, 0x00 }, + { 0x30, 0x01, 0x00 }, + { 0x50, 0x1c, 0x04 }, + { 0xB1, 0x0c, 0x00 }, + { 0xD3, 0x03, 0x03 }, + { 0xE0, 0x01, 0x00 }, + { 0x98, 0x44, 0x04 }, + { 0xB9, 0xff, 0x82 }, + { 0xB7, 0x7777, 0x7273 }, + { 0xB6, 0x07, 0x03 }, + { 0x6B, 0xe0, 0x20 }, + { 0x07, 0xff, 0x70 }, + { 0xBB, 0xff, 0x20 }, + { 0x69, 0xff, 0x40 }, + { 0xBD, 0xffff, 0x17f8 }, + { 0x70, 0xff, 0x15 }, + { 0x7C, 0xff, 0x00 }, + { 0x46, 0xff, 0x1d }, + { 0x1A, 0xffffffff, 0x7fdb7ffe }, + { 0x1B, 0xffffffff, 0x7fdb7ffe }, + { 0x51, 0xff, 0x58 }, + { 0xA2, 0xff, 0xce }, + { 0x33, 0xffff, 0x7fff }, + { 0x4C, 0xffff, 0x0116 }, + { 0x16, 0x1800, 0x0800 }, + { 0x68, 0x1f, 0x07 }, +}; + +static int mt6660_component_setting(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + size_t i = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(component->dev, "%s chip power on failed\n", __func__); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) { + ret = snd_soc_component_update_bits(component, + mt6660_setting_table[i].addr, + mt6660_setting_table[i].mask, + mt6660_setting_table[i].val); + if (ret < 0) { + dev_err(component->dev, "%s update 0x%02x failed\n", + __func__, mt6660_setting_table[i].addr); + return ret; + } + } + + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(component->dev, "%s chip power off failed\n", __func__); + return ret; + } + + return 0; +} + static int mt6660_component_probe(struct snd_soc_component *component) { struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret; dev_dbg(component->dev, "%s\n", __func__); snd_soc_component_init_regmap(component, chip->regmap); - return 0; + ret = mt6660_component_setting(component); + if (ret < 0) + dev_err(chip->dev, "mt6660 component setting failed\n"); + + return ret; } static void mt6660_component_remove(struct snd_soc_component *component) @@ -506,4 +577,4 @@ module_i2c_driver(mt6660_i2c_driver); MODULE_AUTHOR("Jeff Chang <jeff_chang@richtek.com>"); MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); MODULE_LICENSE("GPL"); -MODULE_VERSION("1.0.7_G"); +MODULE_VERSION("1.0.8_G"); diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 287c962ba00d..115706a55577 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -7,6 +7,7 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/device.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> @@ -31,7 +32,7 @@ struct rk3328_codec_priv { struct regmap *regmap; - struct regmap *grf; + struct gpio_desc *mute; struct clk *mclk; struct clk *pclk; unsigned int sclk; @@ -106,16 +107,6 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static void rk3328_analog_output(struct rk3328_codec_priv *rk3328, int mute) -{ - unsigned int val = BIT(17); - - if (mute) - val |= BIT(1); - - regmap_write(rk3328->grf, RK3328_GRF_SOC_CON10, val); -} - static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute) { struct rk3328_codec_priv *rk3328 = @@ -205,7 +196,7 @@ static int rk3328_codec_open_playback(struct rk3328_codec_priv *rk3328) } msleep(rk3328->spk_depop_time); - rk3328_analog_output(rk3328, 1); + gpiod_set_value(rk3328->mute, 0); regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, HPOUTL_GAIN_MASK, OUT_VOLUME); @@ -246,7 +237,7 @@ static int rk3328_codec_close_playback(struct rk3328_codec_priv *rk3328) { size_t i; - rk3328_analog_output(rk3328, 0); + gpiod_set_value(rk3328->mute, 1); regmap_update_bits(rk3328->regmap, HPOUTL_GAIN_CTRL, HPOUTL_GAIN_MASK, 0); @@ -446,7 +437,6 @@ static int rk3328_platform_probe(struct platform_device *pdev) dev_err(&pdev->dev, "missing 'rockchip,grf'\n"); return PTR_ERR(grf); } - rk3328->grf = grf; /* enable i2s_acodec_en */ regmap_write(grf, RK3328_GRF_SOC_CON2, (BIT(14) << 16 | BIT(14))); @@ -458,7 +448,18 @@ static int rk3328_platform_probe(struct platform_device *pdev) rk3328->spk_depop_time = 200; } - rk3328_analog_output(rk3328, 0); + rk3328->mute = gpiod_get_optional(&pdev->dev, "mute", GPIOD_OUT_HIGH); + if (IS_ERR(rk3328->mute)) + return PTR_ERR(rk3328->mute); + /* + * Rock64 is the only supported platform to have widely relied on + * this; if we do happen to come across an old DTB, just leave the + * external mute forced off. + */ + if (!rk3328->mute && of_machine_is_compatible("pine64,rock64")) { + dev_warn(&pdev->dev, "assuming implicit control of GPIO_MUTE; update devicetree if possible\n"); + regmap_write(grf, RK3328_GRF_SOC_CON10, BIT(17) | BIT(1)); + } rk3328->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(rk3328->mclk)) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index a887d5ccb10d..d181c217d835 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -102,6 +102,7 @@ struct pll_calc_map { static const struct pll_calc_map pll_preset_table[] = { {19200000, 4096000, 23, 14, 1, false}, {19200000, 24576000, 3, 30, 3, false}, + {3840000, 24576000, 3, 30, 0, true}, }; static unsigned int find_best_div(unsigned int in, diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h index 31a9643b0afd..6d8ed0377296 100644 --- a/sound/soc/codecs/rl6231.h +++ b/sound/soc/codecs/rl6231.h @@ -10,7 +10,7 @@ #ifndef __RL6231_H__ #define __RL6231_H__ -#define RL6231_PLL_INP_MAX 40000000 +#define RL6231_PLL_INP_MAX 50000000 #define RL6231_PLL_INP_MIN 256000 #define RL6231_PLL_N_MAX 0x1ff #define RL6231_PLL_K_MAX 0x1f diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 66eb55b4ffd4..bb310bc7febd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -444,7 +444,7 @@ static int rt1015_boost_mode_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -457,7 +457,7 @@ static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -497,7 +497,7 @@ static const struct snd_kcontrol_new rt1015_snd_controls[] = { rt1015_boost_mode_get, rt1015_boost_mode_put), SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel), SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0, - rt5518_bypass_boost_get, rt5518_bypass_boost_put), + rt1015_bypass_boost_get, rt1015_bypass_boost_put), }; static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index d930f60cb797..a5a7e46de246 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -507,6 +507,28 @@ static void rt1308_sdw_shutdown(struct snd_pcm_substream *substream, kfree(stream); } +static int rt1308_sdw_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct rt1308_sdw_priv *rt1308 = + snd_soc_component_get_drvdata(component); + + if (tx_mask) + return -EINVAL; + + if (slots > 2) + return -EINVAL; + + rt1308->rx_mask = rx_mask; + rt1308->slots = slots; + /* slot_width is not used since it's irrelevant for SoundWire */ + + return 0; +} + static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -517,7 +539,7 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, struct sdw_port_config port_config; enum sdw_data_direction direction; struct sdw_stream_data *stream; - int retval, port, num_channels; + int retval, port, num_channels, ch_mask; dev_dbg(dai->dev, "%s %s", __func__, dai->name); stream = snd_soc_dai_get_dma_data(dai, substream); @@ -537,13 +559,20 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + if (rt1308->slots) { + num_channels = rt1308->slots; + ch_mask = rt1308->rx_mask; + } else { + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + } + stream_config.frame_rate = params_rate(params); - stream_config.ch_count = params_channels(params); + stream_config.ch_count = num_channels; stream_config.bps = snd_pcm_format_width(params_format(params)); stream_config.direction = direction; - num_channels = params_channels(params); - port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.ch_mask = ch_mask; port_config.num = port; retval = sdw_stream_add_slave(rt1308->sdw_slave, &stream_config, @@ -597,6 +626,7 @@ static const struct snd_soc_dai_ops rt1308_aif_dai_ops = { .hw_free = rt1308_sdw_pcm_hw_free, .set_sdw_stream = rt1308_set_sdw_stream, .shutdown = rt1308_sdw_shutdown, + .set_tdm_slot = rt1308_sdw_set_tdm_slot, }; #define RT1308_STEREO_RATES SNDRV_PCM_RATE_48000 diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index c9341e70d6cf..c5ce75666dcc 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -160,6 +160,8 @@ struct rt1308_sdw_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + int rx_mask; + int slots; }; struct sdw_stream_data { diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index e66d08398f74..89e0f58512fa 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -1604,7 +1604,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); - int pd, idx = -EINVAL; + int pd, idx; pd = rl6231_get_pre_div(rt5659->regmap, RT5659_ADDA_CLK_1, RT5659_I2S_PD1_SFT); diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c new file mode 100644 index 000000000000..a2d1d3ae1e31 --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682-sdw.c -- RT5682 ALSA SoC audio component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Oder Chiou <oder_chiou@realtek.com> +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5682.h" +#include "rt5682-sdw.h" + +static bool rt5682_sdw_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x3000: + case 0x3001: + case 0x3004: + case 0x3005: + case 0x3008: + return true; + default: + return false; + } +} + +const struct regmap_config rt5682_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, + .val_bits = 8, + .max_register = RT5682_I2C_MODE, + .readable_reg = rt5682_sdw_readable_register, + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt5682_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt5682->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt5682->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt5682->hw_init || rt5682->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt5682_io_init(&slave->dev, slave); +} + +static int rt5682_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x4; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +/* Bus clock frequency */ +#define RT5682_CLK_FREQ_9600000HZ 9600000 +#define RT5682_CLK_FREQ_12000000HZ 12000000 +#define RT5682_CLK_FREQ_6000000HZ 6000000 +#define RT5682_CLK_FREQ_4800000HZ 4800000 +#define RT5682_CLK_FREQ_2400000HZ 2400000 +#define RT5682_CLK_FREQ_12288000HZ 12288000 + +static int rt5682_clock_config(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt5682->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT5682_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT5682_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT5682_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT5682_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT5682_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT5682_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt5682->sdw_regmap, 0xe0, value); + regmap_write(rt5682->sdw_regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static int rt5682_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt5682->params, params, sizeof(*params)); + + ret = rt5682_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt5682_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + if (status->control_port & 0x4) { + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} + +static struct sdw_slave_ops rt5682_slave_ops = { + .read_prop = rt5682_read_prop, + .interrupt_callback = rt5682_interrupt_callback, + .update_status = rt5682_update_status, + .bus_config = rt5682_bus_config, +}; + +static int rt5682_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Assign ops */ + slave->ops = &rt5682_slave_ops; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt5682_sdw_regmap); + if (IS_ERR(regmap)) + return -EINVAL; + + rt5682_sdw_init(&slave->dev, regmap, slave); + + return 0; +} + +static int rt5682_sdw_remove(struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + if (rt5682 && rt5682->hw_init) + cancel_delayed_work(&rt5682->jack_detect_work); + + return 0; +} + +static const struct sdw_device_id rt5682_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt5682_id); + +static int __maybe_unused rt5682_dev_suspend(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + if (!rt5682->hw_init) + return 0; + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + + return 0; +} + +static int __maybe_unused rt5682_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt5682->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} + +static const struct dev_pm_ops rt5682_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume) + SET_RUNTIME_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume, NULL) +}; + +static struct sdw_driver rt5682_sdw_driver = { + .driver = { + .name = "rt5682", + .owner = THIS_MODULE, + .pm = &rt5682_pm, + }, + .probe = rt5682_sdw_probe, + .remove = rt5682_sdw_remove, + .ops = &rt5682_slave_ops, + .id_table = rt5682_id, +}; +module_sdw_driver(rt5682_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver SDW"); +MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682-sdw.h b/sound/soc/codecs/rt5682-sdw.h new file mode 100644 index 000000000000..76e6f607066e --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.h @@ -0,0 +1,20 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * rt5682-sdw.h -- RT5682 SDW ALSA SoC audio driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou@realtek.com> + */ + +#ifndef __RT5682_SDW_H__ +#define __RT5682_SDW_H__ + +#define RT5682_SDW_ADDR_L 0x3000 +#define RT5682_SDW_ADDR_H 0x3001 +#define RT5682_SDW_DATA_L 0x3004 +#define RT5682_SDW_DATA_H 0x3005 +#define RT5682_SDW_CMD 0x3008 + +#define RT5682_PROBE_TIMEOUT 2000 + +#endif /* __RT5682_SDW_H__ */ diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index ae6f6121bc1b..c9268a230daa 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -11,13 +11,13 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <linux/regulator/consumer.h> #include <linux/mutex.h> #include <sound/core.h> #include <sound/pcm.h> @@ -31,8 +31,7 @@ #include "rl6231.h" #include "rt5682.h" - -#define RT5682_NUM_SUPPLIES 3 +#include "rt5682-sdw.h" static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { "AVDD", @@ -45,35 +44,15 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, .jd_src = RT5682_JD1, .btndet_delay = 16, -}; - -struct rt5682_priv { - struct snd_soc_component *component; - struct rt5682_platform_data pdata; - struct regmap *regmap; - struct snd_soc_jack *hs_jack; - struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; - struct delayed_work jack_detect_work; - struct delayed_work jd_check_work; - struct mutex calibrate_mutex; - - int sysclk; - int sysclk_src; - int lrck[RT5682_AIFS]; - int bclk[RT5682_AIFS]; - int master[RT5682_AIFS]; - - int pll_src; - int pll_in; - int pll_out; - - int jack_type; + .dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, + {RT5682_PLL2_INTERNAL, 0x8266}, }; static const struct reg_default rt5682_reg[] = { @@ -221,7 +200,7 @@ static const struct reg_default rt5682_reg[] = { {0x0148, 0x0000}, {0x0149, 0x0000}, {0x0150, 0x79a1}, - {0x0151, 0x0000}, + {0x0156, 0xaaaa}, {0x0160, 0x4ec0}, {0x0161, 0x0080}, {0x0162, 0x0200}, @@ -805,10 +784,27 @@ static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); -static void rt5682_reset(struct regmap *regmap) +static const char * const rt5682_dac_select[] = { + "IF1", "SOUND" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_dacl_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_L_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_l_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5682_dacl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682_dacr_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_R_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_r_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5682_dacr_enum); + +static void rt5682_reset(struct rt5682_priv *rt5682) { - regmap_write(regmap, RT5682_RESET, 0); - regmap_write(regmap, RT5682_I2C_MODE, 1); + regmap_write(rt5682->regmap, RT5682_RESET, 0); + if (!rt5682->is_sdw) + regmap_write(rt5682->regmap, RT5682_I2C_MODE, 1); } /** * rt5682_sel_asrc_clk_src - select ASRC clock source for a set of filters @@ -871,6 +867,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) static void rt5682_enable_push_button_irq(struct snd_soc_component *component, bool enable) { + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + if (enable) { snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK, RT5682_SAR_BUTT_DET_EN); @@ -880,8 +878,15 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5682_4BTN_IL_CMD_2, RT5682_4BTN_IL_MASK | RT5682_4BTN_IL_RST_MASK, RT5682_4BTN_IL_EN | RT5682_4BTN_IL_NOR); - snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, - RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_EN); + if (rt5682->is_sdw) + snd_soc_component_update_bits(component, + RT5682_IRQ_CTRL_3, + RT5682_IL_IRQ_MASK | RT5682_IL_IRQ_TYPE_MASK, + RT5682_IL_IRQ_EN | RT5682_IL_IRQ_PUL); + else + snd_soc_component_update_bits(component, + RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK, + RT5682_IL_IRQ_EN); } else { snd_soc_component_update_bits(component, RT5682_IRQ_CTRL_3, RT5682_IL_IRQ_MASK, RT5682_IL_IRQ_DIS); @@ -909,6 +914,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { @@ -917,10 +923,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); usleep_range(15000, 20000); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, RT5682_PWR_CBJ); @@ -951,8 +957,13 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); + else + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -999,62 +1010,69 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!rt5682->is_sdw) { + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } - switch (rt5682->pdata.jd_src) { - case RT5682_JD1: - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, - RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); - snd_soc_component_write(component, RT5682_CBJ_CTRL_1, 0xd042); - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_3, - RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); - snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); - regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, - RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + switch (rt5682->pdata.jd_src) { + case RT5682_JD1: + snd_soc_component_update_bits(component, + RT5682_CBJ_CTRL_2, RT5682_EXT_JD_SRC, + RT5682_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5682_CBJ_CTRL_1, + 0xd042); + snd_soc_component_update_bits(component, + RT5682_CBJ_CTRL_3, RT5682_CBJ_IN_BUF_EN, + RT5682_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_1, RT5682_SAR_POW_MASK, + RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, + RT5682_GP1_PIN_MASK, RT5682_GP1_PIN_IRQ); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, RT5682_POW_IRQ | RT5682_POW_JDH | RT5682_POW_ANA, RT5682_POW_IRQ | RT5682_POW_JDH | RT5682_POW_ANA); - regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, - RT5682_PWR_JDH | RT5682_PWR_JDL, - RT5682_PWR_JDH | RT5682_PWR_JDL); - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, - RT5682_JD1_EN | RT5682_JD1_POL_NOR); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7, - 0x7f7f, (rt5682->pdata.btndet_delay << 8 | - rt5682->pdata.btndet_delay)); - mod_delayed_work(system_power_efficient_wq, - &rt5682->jack_detect_work, msecs_to_jiffies(250)); - break; + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH | RT5682_PWR_JDL, + RT5682_PWR_JDH | RT5682_PWR_JDL); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_POL_MASK, + RT5682_JD1_EN | RT5682_JD1_POL_NOR); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_4, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_5, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_6, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + regmap_update_bits(rt5682->regmap, RT5682_4BTN_IL_CMD_7, + 0x7f7f, (rt5682->pdata.btndet_delay << 8 | + rt5682->pdata.btndet_delay)); + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, + msecs_to_jiffies(250)); + break; - case RT5682_JD_NULL: - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - break; + case RT5682_JD_NULL: + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + break; - default: - dev_warn(component->dev, "Wrong JD source\n"); - break; + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } } return 0; @@ -1134,11 +1152,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); - if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3)) - schedule_delayed_work(&rt5682->jd_check_work, 0); - else - cancel_delayed_work_sync(&rt5682->jd_check_work); + if (!rt5682->is_sdw) { + if (rt5682->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5682->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5682->jd_check_work); + } mutex_unlock(&rt5682->calibrate_mutex); } @@ -1146,7 +1166,7 @@ static void rt5682_jack_detect_handler(struct work_struct *work) static const struct snd_kcontrol_new rt5682_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, - RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 87, 0, dac_vol_tlv), /* IN Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, @@ -1177,11 +1197,11 @@ static int rt5682_div_sel(struct rt5682_priv *rt5682, } for (i = 0; i < size - 1; i++) { - pr_info("div[%d]=%d\n", i, div[i]); + dev_dbg(rt5682->component->dev, "div[%d]=%d\n", i, div[i]); if (target * div[i] == rt5682->sysclk) return i; if (target * div[i + 1] > rt5682->sysclk) { - pr_err("can't find div for sysclk %d\n", + dev_dbg(rt5682->component->dev, "can't find div for sysclk %d\n", rt5682->sysclk); return i; } @@ -1211,10 +1231,13 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - int idx = -EINVAL; + int idx = -EINVAL, dmic_clk_rate = 3072000; static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; - idx = rt5682_div_sel(rt5682, 1500000, div, ARRAY_SIZE(div)); + if (rt5682->pdata.dmic_clk_rate) + dmic_clk_rate = rt5682->pdata.dmic_clk_rate; + + idx = rt5682_div_sel(rt5682, dmic_clk_rate, div, ARRAY_SIZE(div)); snd_soc_component_update_bits(component, RT5682_DMIC_CTRL_1, RT5682_DMIC_CLK_MASK, idx << RT5682_DMIC_CLK_SFT); @@ -1232,6 +1255,9 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + if (rt5682->is_sdw) + return 0; + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & RT5682_GP4_PIN_MASK; if (w->shift == RT5682_PWR_ADC_S1F_BIT && @@ -1278,6 +1304,21 @@ static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, return 0; } +static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL2) + return 1; + else + return 0; +} + static int is_using_asrc(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -1503,10 +1544,18 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, static int set_dmic_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int delay = 50; + + if (rt5682->pdata.dmic_delay) + delay = rt5682->pdata.dmic_delay; + switch (event) { case SND_SOC_DAPM_POST_PMU: /*Add delay to avoid pop noise*/ - msleep(150); + msleep(delay); break; default: @@ -1516,7 +1565,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static int rt5655_set_verf(struct snd_soc_dapm_widget *w, +static int rt5682_set_verf(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = @@ -1592,9 +1641,12 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, - 0, NULL, 0), + 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1686,6 +1738,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SOUND DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SOUND DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), /* Digital Interface Select */ SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, @@ -1702,12 +1756,19 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, &rt5682_adcdat_pin_ctrl), + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_l_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_r_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SDWRX", "SDW Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SDWTX", "SDW Capture", 0, SND_SOC_NOPM, 0, 0), /* Output Side */ /* DAC mixer before sound effect */ @@ -1776,7 +1837,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*PLL*/ {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, @@ -1860,8 +1925,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, - {"IF1_ADC Mux", NULL, "I2S1"}, {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, {"AIF1TX", NULL, "ADCDAT Mux"}, {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, @@ -1870,6 +1935,10 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, {"AIF2TX", NULL, "ADCDAT Mux"}, + {"SDWTX", NULL, "PLL2B"}, + {"SDWTX", NULL, "PLL2F"}, + {"SDWTX", NULL, "ADCDAT Mux"}, + {"IF1 DAC1 L", NULL, "AIF1RX"}, {"IF1 DAC1 L", NULL, "I2S1"}, {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, @@ -1877,10 +1946,24 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1 DAC1 R", NULL, "I2S1"}, {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "SDWRX"}, + {"SOUND DAC L", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "PLL2B"}, + {"SOUND DAC L", NULL, "PLL2F"}, + {"SOUND DAC R", NULL, "SDWRX"}, + {"SOUND DAC R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC R", NULL, "PLL2B"}, + {"SOUND DAC R", NULL, "PLL2F"}, + + {"DAC L Mux", "IF1", "IF1 DAC1 L"}, + {"DAC L Mux", "SOUND", "SOUND DAC L"}, + {"DAC R Mux", "IF1", "IF1 DAC1 R"}, + {"DAC R Mux", "SOUND", "SOUND DAC R"}, + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, - {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXL", "DAC1 Switch", "DAC L Mux"}, {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, - {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + {"DAC1 MIXR", "DAC1 Switch", "DAC R Mux"}, {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, @@ -2033,8 +2116,10 @@ static int rt5682_hw_params(struct snd_pcm_substream *substream, RT5682_I2S1_DL_MASK, len_1); if (rt5682->master[RT5682_AIF1]) { snd_soc_component_update_bits(component, - RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, - pre_div << RT5682_I2S_M_DIV_SFT); + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK | + RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); } if (params_channels(params) == 1) /* mono mode */ snd_soc_component_update_bits(component, @@ -2207,61 +2292,157 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, unsigned int freq_out) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct rl6231_pll_code pll_code; + struct rl6231_pll_code pll_code, pll2f_code, pll2b_code; + unsigned int pll2_fout1; int ret; - if (source == rt5682->pll_src && freq_in == rt5682->pll_in && - freq_out == rt5682->pll_out) + if (source == rt5682->pll_src[pll_id] && + freq_in == rt5682->pll_in[pll_id] && + freq_out == rt5682->pll_out[pll_id]) return 0; if (!freq_in || !freq_out) { dev_dbg(component->dev, "PLL disabled\n"); - rt5682->pll_in = 0; - rt5682->pll_out = 0; + rt5682->pll_in[pll_id] = 0; + rt5682->pll_out[pll_id] = 0; snd_soc_component_update_bits(component, RT5682_GLB_CLK, RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); return 0; } - switch (source) { - case RT5682_PLL1_S_MCLK: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); - break; - case RT5682_PLL1_S_BCLK1: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); - break; - default: - dev_err(component->dev, "Unknown PLL Source %d\n", source); - return -EINVAL; - } + if (pll_id == RT5682_PLL2) { + switch (source) { + case RT5682_PLL2_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK, + RT5682_PLL2_SRC_MCLK); + break; + default: + dev_err(component->dev, "Unknown PLL2 Source %d\n", + source); + return -EINVAL; + } - ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); - if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); - return ret; + /** + * PLL2 concatenates 2 PLL units. + * We suggest the Fout of the front PLL is 3.84MHz. + */ + pll2_fout1 = 3840000; + ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + freq_in, pll2_fout1, + pll2f_code.m_bp, + (pll2f_code.m_bp ? 0 : pll2f_code.m_code), + pll2f_code.n_code, pll2f_code.k_code); + + ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + pll2_fout1); + return ret; + } + dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + pll2_fout1, freq_out, + pll2b_code.m_bp, + (pll2b_code.m_bp ? 0 : pll2b_code.m_code), + pll2b_code.n_code, pll2b_code.k_code); + + snd_soc_component_write(component, RT5682_PLL2_CTRL_1, + pll2f_code.k_code << RT5682_PLL2F_K_SFT | + pll2b_code.k_code << RT5682_PLL2B_K_SFT | + pll2b_code.m_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_2, + pll2f_code.m_code << RT5682_PLL2F_M_SFT | + pll2b_code.n_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_3, + pll2f_code.n_code << RT5682_PLL2F_N_SFT); + snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4, + RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf, + (pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT | + (pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT | + 0xf); + } else { + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL1 Source %d\n", + source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); } - dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), - pll_code.n_code, pll_code.k_code); + rt5682->pll_in[pll_id] = freq_in; + rt5682->pll_out[pll_id] = freq_out; + rt5682->pll_src[pll_id] = source; - snd_soc_component_write(component, RT5682_PLL_CTRL_1, - pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); - snd_soc_component_write(component, RT5682_PLL_CTRL_2, - (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | - pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + return 0; +} - rt5682->pll_in = freq_in; - rt5682->pll_out = freq_out; - rt5682->pll_src = source; +static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + + rt5682->bclk[dai->id] = ratio; + + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } return 0; } -static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) { struct snd_soc_component *component = dai->component; struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); @@ -2280,7 +2461,7 @@ static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) RT5682_I2S2_BCLK_MS2_32); break; default: - dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); return -EINVAL; } @@ -2319,12 +2500,392 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, return 0; } +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_PLL2_FOUT 24576000 +#define CLK_48 48000 + +static bool rt5682_clk_check(struct rt5682_priv *rt5682) +{ + if (!rt5682->master[RT5682_AIF1]) { + dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n"); + return false; + } + return true; +} + +static int rt5682_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + if (!rt5682->jack_type) + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return 0; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct clk *parent_clk; + const char * const clk_name = __clk_get_name(hw->clk); + int pre_div; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * It's a temporary limitation. Only accept to set wclk rate to 48kHz. + * It will force wclk to 48kHz even it's not. + */ + if (rate != CLK_48) { + dev_warn(component->dev, "clk %s only support %d Hz output\n", + clk_name, CLK_48); + rate = CLK_48; + } + + /* + * To achieve the rate conversion from 48MHz to 48kHz, PLL2 is needed. + */ + rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CLK_PLL2_FIN, CLK_PLL2_FOUT); + + rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0, + CLK_PLL2_FOUT, SND_SOC_CLOCK_IN); + + pre_div = rl6231_get_clk_info(rt5682->sysclk, rate); + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); + + return 0; +} + +static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + unsigned int bclks_per_wclk; + + snd_soc_component_read(component, RT5682_TDM_TCON_CTRL, + &bclks_per_wclk); + + switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { + case RT5682_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dai *dai = NULL; + unsigned long factor; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + factor = rt5682_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; + } + + return rt5682_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { + [RT5682_DAI_WCLK_IDX] = { + .prepare = rt5682_wclk_prepare, + .unprepare = rt5682_wclk_unprepare, + .recalc_rate = rt5682_wclk_recalc_rate, + .round_rate = rt5682_wclk_round_rate, + .set_rate = rt5682_wclk_set_rate, + }, + [RT5682_DAI_BCLK_IDX] = { + .recalc_rate = rt5682_bclk_recalc_rate, + .round_rate = rt5682_bclk_round_rate, + .set_rate = rt5682_bclk_set_rate, + }, +}; + +static int rt5682_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rt5682_platform_data *pdata = &rt5682->pdata; + struct clk_init_data init; + struct clk *dai_clk; + struct clk_lookup *dai_clk_lookup; + struct clk_hw *dai_clk_hw; + const char *parent_name; + int i, ret; + + for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { + dai_clk_hw = &rt5682->dai_clks_hw[i]; + + switch (i) { + case RT5682_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682->mclk) { + parent_name = __clk_get_name(rt5682->mclk); + init.parent_names = &parent_name; + init.num_parents = 1; + } else { + init.parent_names = NULL; + init.num_parents = 0; + } + break; + case RT5682_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + parent_name = __clk_get_name( + rt5682->dai_clks[RT5682_DAI_WCLK_IDX]); + init.parent_names = &parent_name; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + ret = -EINVAL; + goto err; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + dai_clk = devm_clk_register(dev, dai_clk_hw); + if (IS_ERR(dai_clk)) { + dev_warn(dev, "Failed to register %s: %ld\n", + init.name, PTR_ERR(dai_clk)); + ret = PTR_ERR(dai_clk); + goto err; + } + rt5682->dai_clks[i] = dai_clk; + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + dai_clk_lookup = clkdev_create(dai_clk, init.name, + "%s", dev_name(dev)); + if (!dai_clk_lookup) { + ret = -ENOMEM; + goto err; + } else { + rt5682->dai_clks_lookup[i] = dai_clk_lookup; + } + } + } + + return 0; + +err: + do { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } while (i-- > 0); + + return ret; +} +#endif /* CONFIG_COMMON_CLK */ + static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_slave *slave; + unsigned long time; +#ifdef CONFIG_COMMON_CLK + int ret; +#endif rt5682->component = component; + if (rt5682->is_sdw) { + slave = rt5682->slave; + time = wait_for_completion_timeout( + &slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + } else { +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } else { + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + } + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif + } + return 0; } @@ -2332,7 +2893,16 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - rt5682_reset(rt5682->regmap); +#ifdef CONFIG_COMMON_CLK + int i; + + for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } +#endif + + rt5682_reset(rt5682); } #ifdef CONFIG_PM @@ -2369,14 +2939,203 @@ static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, .set_tdm_slot = rt5682_set_tdm_slot, + .set_bclk_ratio = rt5682_set_bclk1_ratio, }; static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, - .set_bclk_ratio = rt5682_set_bclk_ratio, + .set_bclk_ratio = rt5682_set_bclk2_ratio, }; +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt5682_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val_p = 0, val_c = 0, osr_p = 0, osr_c = 0; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -ENOMEM; + + if (!rt5682->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt5682->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + switch (params_rate(params)) { + case 48000: + val_p = RT5682_SDW_REF_1_48K; + val_c = RT5682_SDW_REF_2_48K; + break; + case 96000: + val_p = RT5682_SDW_REF_1_96K; + val_c = RT5682_SDW_REF_2_96K; + break; + case 192000: + val_p = RT5682_SDW_REF_1_192K; + val_c = RT5682_SDW_REF_2_192K; + break; + case 32000: + val_p = RT5682_SDW_REF_1_32K; + val_c = RT5682_SDW_REF_2_32K; + break; + case 24000: + val_p = RT5682_SDW_REF_1_24K; + val_c = RT5682_SDW_REF_2_24K; + break; + case 16000: + val_p = RT5682_SDW_REF_1_16K; + val_c = RT5682_SDW_REF_2_16K; + break; + case 12000: + val_p = RT5682_SDW_REF_1_12K; + val_c = RT5682_SDW_REF_2_12K; + break; + case 8000: + val_p = RT5682_SDW_REF_1_8K; + val_c = RT5682_SDW_REF_2_8K; + break; + case 44100: + val_p = RT5682_SDW_REF_1_44K; + val_c = RT5682_SDW_REF_2_44K; + break; + case 88200: + val_p = RT5682_SDW_REF_1_88K; + val_c = RT5682_SDW_REF_2_88K; + break; + case 176400: + val_p = RT5682_SDW_REF_1_176K; + val_c = RT5682_SDW_REF_2_176K; + break; + case 22050: + val_p = RT5682_SDW_REF_1_22K; + val_c = RT5682_SDW_REF_2_22K; + break; + case 11025: + val_p = RT5682_SDW_REF_1_11K; + val_c = RT5682_SDW_REF_2_11K; + break; + default: + return -EINVAL; + } + + if (params_rate(params) <= 48000) { + osr_p = RT5682_DAC_OSR_D_8; + osr_c = RT5682_ADC_OSR_D_8; + } else if (params_rate(params) <= 96000) { + osr_p = RT5682_DAC_OSR_D_4; + osr_c = RT5682_ADC_OSR_D_4; + } else { + osr_p = RT5682_DAC_OSR_D_2; + osr_c = RT5682_ADC_OSR_D_2; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_1_MASK, val_p); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_DAC_OSR_MASK, osr_p); + } else { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_2_MASK, val_c); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_ADC_OSR_MASK, osr_c); + } + + return retval; +} + +static int rt5682_sdw_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt5682->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt5682->slave, stream->sdw_stream); + return 0; +} + +static struct snd_soc_dai_ops rt5682_sdw_ops = { + .hw_params = rt5682_sdw_hw_params, + .hw_free = rt5682_sdw_hw_free, + .set_sdw_stream = rt5682_set_sdw_stream, + .shutdown = rt5682_sdw_shutdown, +}; +#endif + static struct snd_soc_dai_driver rt5682_dai[] = { { .name = "rt5682-aif1", @@ -2409,6 +3168,27 @@ static struct snd_soc_dai_driver rt5682_dai[] = { }, .ops = &rt5682_aif2_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) + { + .name = "rt5682-sdw", + .id = RT5682_SDW, + .playback = { + .stream_name = "SDW Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "SDW Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_sdw_ops, + }, +#endif }; static const struct snd_soc_component_driver soc_component_dev_rt5682 = { @@ -2461,10 +3241,21 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) &rt5682->pdata.jd_src); device_property_read_u32(dev, "realtek,btndet-delay", &rt5682->pdata.btndet_delay); + device_property_read_u32(dev, "realtek,dmic-clk-rate-hz", + &rt5682->pdata.dmic_clk_rate); + device_property_read_u32(dev, "realtek,dmic-delay-ms", + &rt5682->pdata.dmic_delay); rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); + if (device_property_read_string_array(dev, "clock-output-names", + rt5682->pdata.dai_clk_names, + RT5682_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], + rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + return 0; } @@ -2474,7 +3265,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); @@ -2520,6 +3311,221 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) } +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +static int rt5682_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int data_l, data_h; + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 0); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_H, &data_h); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_L, &data_l); + + *val = (data_h << 8) | data_l; + + dev_vdbg(dev, "[%s] %04x => %04x\n", __func__, reg, *val); + + return 0; +} + +static int rt5682_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 1); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_H, (val >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_L, (val & 0xff)); + + dev_vdbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + + return 0; +} + +static const struct regmap_config rt5682_sdw_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_read = true, + .use_single_write = true, + .reg_read = rt5682_sdw_read, + .reg_write = rt5682_sdw_write, +}; + +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682; + int ret; + + rt5682 = devm_kzalloc(dev, sizeof(*rt5682), GFP_KERNEL); + if (!rt5682) + return -ENOMEM; + + dev_set_drvdata(dev, rt5682); + rt5682->slave = slave; + rt5682->sdw_regmap = regmap; + rt5682->is_sdw = true; + + rt5682->regmap = devm_regmap_init(dev, NULL, dev, &rt5682_sdw_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt5682->hw_init = false; + rt5682->first_hw_init = false; + + mutex_init(&rt5682->calibrate_mutex); + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_sdw_init); + +int rt5682_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + int ret = 0; + unsigned int val; + + if (rt5682->hw_init) + return 0; + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt5682->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + rt5682_reset(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_only(rt5682->regmap, false); + regcache_cache_bypass(rt5682->regmap, true); + } + + rt5682_calibrate(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_bypass(rt5682->regmap, false); + regcache_mark_dirty(rt5682->regmap); + regcache_sync(rt5682->regmap); + + /* volatile registers */ + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + + goto reinit; + } + + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); + regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, + RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + + /* Soundwire */ + regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_1, 0x1700); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_2, 0x0006); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_3, 0x2600); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_4, 0x0c8f); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_2, 0x3000); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_3, 0x4000); + regmap_update_bits(rt5682->regmap, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK | RT5682_PLL2_SRC_MASK, + RT5682_SCLK_SRC_PLL2 | RT5682_PLL2_SRC_SDW); + + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH, RT5682_PWR_JDH); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_IRQ_MASK, + RT5682_JD1_EN | RT5682_JD1_IRQ_PUL); + +reinit: + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + /* Mark Slave initialization complete */ + rt5682->hw_init = true; + rt5682->first_hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_io_init); +#endif + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2586,7 +3592,7 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); @@ -2651,6 +3657,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); + regmap_update_bits(rt5682->regmap, RT5682_DMIC_CTRL_1, + RT5682_FIFO_CLK_DIV_MASK, RT5682_FIFO_CLK_DIV_2); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); @@ -2676,7 +3684,7 @@ static void rt5682_i2c_shutdown(struct i2c_client *client) { struct rt5682_priv *rt5682 = i2c_get_clientdata(client); - rt5682_reset(rt5682->regmap); + rt5682_reset(rt5682); } #ifdef CONFIG_OF diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 18faaa2a49a0..0baeece84ec4 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -10,6 +10,12 @@ #define __RT5682_H__ #include <sound/rt5682.h> +#include <linux/regulator/consumer.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> #define DEVICE_ID 0x6530 @@ -177,7 +183,7 @@ #define RT5682_TEST_MODE_CTRL_4 0x0148 #define RT5682_TEST_MODE_CTRL_5 0x0149 #define RT5682_PLL1_INTERNAL 0x0150 -#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_PLL2_INTERNAL 0x0156 #define RT5682_STO_NG2_CTRL_1 0x0160 #define RT5682_STO_NG2_CTRL_2 0x0161 #define RT5682_STO_NG2_CTRL_3 0x0162 @@ -651,6 +657,8 @@ #define RT5682_DMIC_1_EN_SFT 15 #define RT5682_DMIC_1_DIS (0x0 << 15) #define RT5682_DMIC_1_EN (0x1 << 15) +#define RT5682_FIFO_CLK_DIV_MASK (0x7 << 12) +#define RT5682_FIFO_CLK_DIV_2 (0x1 << 12) #define RT5682_DMIC_1_DP_MASK (0x3 << 4) #define RT5682_DMIC_1_DP_SFT 4 #define RT5682_DMIC_1_DP_GPIO2 (0x0 << 4) @@ -738,7 +746,7 @@ #define RT5682_ADC_OSR_D_24 (0x7 << 12) #define RT5682_ADC_OSR_D_32 (0x8 << 12) #define RT5682_ADC_OSR_D_48 (0x9 << 12) -#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 8) #define RT5682_I2S_M_DIV_SFT 8 #define RT5682_I2S_M_D_1 (0x0 << 8) #define RT5682_I2S_M_D_2 (0x1 << 8) @@ -820,6 +828,12 @@ #define RT5682_TDM_DF_PCM_B (0x3 << 11) #define RT5682_TDM_DF_PCM_A_N (0x6 << 11) #define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_BCLK_MS1_MASK (0x3 << 9) +#define RT5682_TDM_BCLK_MS1_SFT 9 +#define RT5682_TDM_BCLK_MS1_32 (0x0 << 9) +#define RT5682_TDM_BCLK_MS1_64 (0x1 << 9) +#define RT5682_TDM_BCLK_MS1_128 (0x2 << 9) +#define RT5682_TDM_BCLK_MS1_256 (0x3 << 9) #define RT5682_TDM_CL_MASK (0x3 << 4) #define RT5682_TDM_CL_16 (0x0 << 4) #define RT5682_TDM_CL_20 (0x1 << 4) @@ -835,8 +849,8 @@ #define RT5682_TDM_M_LP_INV (0x1 << 1) #define RT5682_TDM_MS_MASK (0x1 << 0) #define RT5682_TDM_MS_SFT 0 -#define RT5682_TDM_MS_M (0x0 << 0) -#define RT5682_TDM_MS_S (0x1 << 0) +#define RT5682_TDM_MS_S (0x0 << 0) +#define RT5682_TDM_MS_M (0x1 << 0) /* Global Clock Control (0x0080) */ #define RT5682_SCLK_SRC_MASK (0x7 << 13) @@ -1049,6 +1063,28 @@ #define RT5682_PWR_CLK1M_PD (0x0 << 8) #define RT5682_PWR_CLK1M_PU (0x1 << 8) +/* PLL2 M/N/K Code Control 1 (0x009b) */ +#define RT5682_PLL2F_K_MASK (0x1f << 8) +#define RT5682_PLL2F_K_SFT 8 +#define RT5682_PLL2B_K_MASK (0xf << 4) +#define RT5682_PLL2B_K_SFT 4 +#define RT5682_PLL2B_M_MASK (0xf << 0) + +/* PLL2 M/N/K Code Control 2 (0x009c) */ +#define RT5682_PLL2F_M_MASK (0x3f << 8) +#define RT5682_PLL2F_M_SFT 8 +#define RT5682_PLL2B_N_MASK (0x3f << 0) + +/* PLL2 M/N/K Code Control 2 (0x009d) */ +#define RT5682_PLL2F_N_MASK (0x7f << 8) +#define RT5682_PLL2F_N_SFT 8 + +/* PLL2 M/N/K Code Control 2 (0x009e) */ +#define RT5682_PLL2B_M_BP_MASK (0x1 << 11) +#define RT5682_PLL2B_M_BP_SFT 11 +#define RT5682_PLL2F_M_BP_MASK (0x1 << 7) +#define RT5682_PLL2F_M_BP_SFT 7 + /* RC Clock Control (0x009f) */ #define RT5682_POW_IRQ (0x1 << 15) #define RT5682_POW_JDH (0x1 << 14) @@ -1091,11 +1127,17 @@ #define RT5682_JD1_POL_MASK (0x1 << 13) #define RT5682_JD1_POL_NOR (0x0 << 13) #define RT5682_JD1_POL_INV (0x1 << 13) +#define RT5682_JD1_IRQ_MASK (0x1 << 10) +#define RT5682_JD1_IRQ_LEV (0x0 << 10) +#define RT5682_JD1_IRQ_PUL (0x1 << 10) /* IRQ Control 3 (0x00b8) */ #define RT5682_IL_IRQ_MASK (0x1 << 7) #define RT5682_IL_IRQ_DIS (0x0 << 7) #define RT5682_IL_IRQ_EN (0x1 << 7) +#define RT5682_IL_IRQ_TYPE_MASK (0x1 << 4) +#define RT5682_IL_IRQ_LEV (0x0 << 4) +#define RT5682_IL_IRQ_PUL (0x1 << 4) /* GPIO Control 1 (0x00c0) */ #define RT5682_GP1_PIN_MASK (0x3 << 14) @@ -1309,11 +1351,19 @@ enum { RT5682_PLL1_S_MCLK, RT5682_PLL1_S_BCLK1, RT5682_PLL1_S_RCCLK, + RT5682_PLL2_S_MCLK, +}; + +enum { + RT5682_PLL1, + RT5682_PLL2, + RT5682_PLLS, }; enum { RT5682_AIF1, RT5682_AIF2, + RT5682_SDW, RT5682_AIFS }; @@ -1329,7 +1379,49 @@ enum { RT5682_CLK_SEL_I2S2_ASRC, }; +#define RT5682_NUM_SUPPLIES 3 + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct regmap *sdw_regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + bool is_sdw; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; + + int jack_type; +}; + int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, unsigned int filter_mask, unsigned int clk_src); +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave); +int rt5682_io_init(struct device *dev, struct sdw_slave *slave); #endif /* __RT5682_H__ */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index be52886a5edb..7fae88655a0f 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,12 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; +}; + +enum tas256x_model { + TAS2562, + TAS2563, }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -383,21 +407,81 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_POST_PMU: - dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_MUTE); + if (ret) + goto end; break; case SND_SOC_DAPM_PRE_PMD: - dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_SHUTDOWN); + if (ret) + goto end; break; default: - break; + dev_err(tas2562->dev, "Not supported evevt\n"); + return -EINVAL; } +end: + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; return 0; } +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -409,14 +493,24 @@ static const struct snd_kcontrol_new vsense_switch = 1, 1); static const struct snd_kcontrol_new tas2562_snd_controls[] = { - SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, + SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 1, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), - SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), @@ -431,7 +525,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = { {"ASI1 Sel", "Left", "ASI1"}, {"ASI1 Sel", "Right", "ASI1"}, {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, - { "DAC", NULL, "DAC IN" }, + { "DAC", NULL, "ASI1 Sel" }, { "OUT", NULL, "DAC" }, {"ISENSE", "Switch", "IMON"}, {"VSENSE", "Switch", "VMON"}, @@ -472,6 +566,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2562_FORMATS, }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, .ops = &tas2562_speaker_dai_ops, }, }; @@ -495,6 +596,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { @@ -564,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client, } static const struct i2c_device_id tas2562_id[] = { - { "tas2562", 0 }, + { "tas2562", TAS2562 }, + { "tas2563", TAS2563 }, { } }; MODULE_DEVICE_TABLE(i2c, tas2562_id); static const struct of_device_id tas2562_of_match[] = { { .compatible = "ti,tas2562", }, + { .compatible = "ti,tas2563", }, { }, }; MODULE_DEVICE_TABLE(of, tas2562_of_match); diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 62e659ab786d..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,12 +35,14 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -#define TAS2562_MODE_MASK 0x3 +#define TAS2562_MODE_MASK GENMASK(1,0) #define TAS2562_ACTIVE 0x0 #define TAS2562_MUTE 0x1 #define TAS2562_SHUTDOWN 0x2 @@ -73,8 +75,8 @@ #define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) #define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) -#define TAS2562_VSENSE_POWER_EN BIT(2) -#define TAS2562_ISENSE_POWER_EN BIT(3) +#define TAS2562_VSENSE_POWER_EN 2 +#define TAS2562_ISENSE_POWER_EN 3 #define TAS2562_TDM_CFG5_VSNS_EN BIT(6) #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c new file mode 100644 index 000000000000..38897568ee96 --- /dev/null +++ b/sound/soc/codecs/tlv320adcx140.c @@ -0,0 +1,920 @@ +// SPDX-License-Identifier: GPL-2.0 +// TLV320ADCX140 Sound driver +// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> +#include <linux/acpi.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tlv320adcx140.h" + +struct adcx140_priv { + struct snd_soc_component *component; + struct regulator *supply_areg; + struct gpio_desc *gpio_reset; + struct regmap *regmap; + struct device *dev; + + int micbias_vg; + + unsigned int dai_fmt; + unsigned int tdm_delay; + unsigned int slot_width; +}; + +static const struct reg_default adcx140_reg_defaults[] = { + { ADCX140_PAGE_SELECT, 0x00 }, + { ADCX140_SW_RESET, 0x00 }, + { ADCX140_SLEEP_CFG, 0x00 }, + { ADCX140_SHDN_CFG, 0x05 }, + { ADCX140_ASI_CFG0, 0x30 }, + { ADCX140_ASI_CFG1, 0x00 }, + { ADCX140_ASI_CFG2, 0x00 }, + { ADCX140_ASI_CH1, 0x00 }, + { ADCX140_ASI_CH2, 0x01 }, + { ADCX140_ASI_CH3, 0x02 }, + { ADCX140_ASI_CH4, 0x03 }, + { ADCX140_ASI_CH5, 0x04 }, + { ADCX140_ASI_CH6, 0x05 }, + { ADCX140_ASI_CH7, 0x06 }, + { ADCX140_ASI_CH8, 0x07 }, + { ADCX140_MST_CFG0, 0x02 }, + { ADCX140_MST_CFG1, 0x48 }, + { ADCX140_ASI_STS, 0xff }, + { ADCX140_CLK_SRC, 0x10 }, + { ADCX140_PDMCLK_CFG, 0x40 }, + { ADCX140_PDM_CFG, 0x00 }, + { ADCX140_GPIO_CFG0, 0x22 }, + { ADCX140_GPO_CFG1, 0x00 }, + { ADCX140_GPO_CFG2, 0x00 }, + { ADCX140_GPO_CFG3, 0x00 }, + { ADCX140_GPO_CFG4, 0x00 }, + { ADCX140_GPO_VAL, 0x00 }, + { ADCX140_GPIO_MON, 0x00 }, + { ADCX140_GPI_CFG0, 0x00 }, + { ADCX140_GPI_CFG1, 0x00 }, + { ADCX140_GPI_MON, 0x00 }, + { ADCX140_INT_CFG, 0x00 }, + { ADCX140_INT_MASK0, 0xff }, + { ADCX140_INT_LTCH0, 0x00 }, + { ADCX140_BIAS_CFG, 0x00 }, + { ADCX140_CH1_CFG0, 0x00 }, + { ADCX140_CH1_CFG1, 0x00 }, + { ADCX140_CH1_CFG2, 0xc9 }, + { ADCX140_CH1_CFG3, 0x80 }, + { ADCX140_CH1_CFG4, 0x00 }, + { ADCX140_CH2_CFG0, 0x00 }, + { ADCX140_CH2_CFG1, 0x00 }, + { ADCX140_CH2_CFG2, 0xc9 }, + { ADCX140_CH2_CFG3, 0x80 }, + { ADCX140_CH2_CFG4, 0x00 }, + { ADCX140_CH3_CFG0, 0x00 }, + { ADCX140_CH3_CFG1, 0x00 }, + { ADCX140_CH3_CFG2, 0xc9 }, + { ADCX140_CH3_CFG3, 0x80 }, + { ADCX140_CH3_CFG4, 0x00 }, + { ADCX140_CH4_CFG0, 0x00 }, + { ADCX140_CH4_CFG1, 0x00 }, + { ADCX140_CH4_CFG2, 0xc9 }, + { ADCX140_CH4_CFG3, 0x80 }, + { ADCX140_CH4_CFG4, 0x00 }, + { ADCX140_CH5_CFG2, 0xc9 }, + { ADCX140_CH5_CFG3, 0x80 }, + { ADCX140_CH5_CFG4, 0x00 }, + { ADCX140_CH6_CFG2, 0xc9 }, + { ADCX140_CH6_CFG3, 0x80 }, + { ADCX140_CH6_CFG4, 0x00 }, + { ADCX140_CH7_CFG2, 0xc9 }, + { ADCX140_CH7_CFG3, 0x80 }, + { ADCX140_CH7_CFG4, 0x00 }, + { ADCX140_CH8_CFG2, 0xc9 }, + { ADCX140_CH8_CFG3, 0x80 }, + { ADCX140_CH8_CFG4, 0x00 }, + { ADCX140_DSP_CFG0, 0x01 }, + { ADCX140_DSP_CFG1, 0x40 }, + { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_AGC_CFG0, 0xe7 }, + { ADCX140_IN_CH_EN, 0xf0 }, + { ADCX140_ASI_OUT_CH_EN, 0x00 }, + { ADCX140_PWR_CFG, 0x00 }, + { ADCX140_DEV_STS0, 0x00 }, + { ADCX140_DEV_STS1, 0x80 }, +}; + +static const struct regmap_range_cfg adcx140_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = ADCX140_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static bool adcx140_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADCX140_SW_RESET: + case ADCX140_DEV_STS0: + case ADCX140_DEV_STS1: + case ADCX140_ASI_STS: + return true; + default: + return false; + } +} + +static const struct regmap_config adcx140_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adcx140_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adcx140_reg_defaults), + .cache_type = REGCACHE_FLAT, + .ranges = adcx140_ranges, + .num_ranges = ARRAY_SIZE(adcx140_ranges), + .max_register = 12 * 128, + .volatile_reg = adcx140_volatile, +}; + +/* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); + +/* ADC gain. From 0 to 42 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); + +/* DRE Level. From -12 dB to -66 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0); +/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0); + +/* AGC Level. From -6 dB to -36 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); +/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); + +static const char * const decimation_filter_text[] = { + "Linear Phase", "Low Latency", "Ultra-low Latency" +}; + +static SOC_ENUM_SINGLE_DECL(decimation_filter_enum, ADCX140_DSP_CFG0, 4, + decimation_filter_text); + +static const struct snd_kcontrol_new decimation_filter_controls[] = { + SOC_DAPM_ENUM("Decimation Filter", decimation_filter_enum), +}; + +static const char * const resistor_text[] = { + "2.5 kOhm", "10 kOhm", "20 kOhm" +}; + +static SOC_ENUM_SINGLE_DECL(in1_resistor_enum, ADCX140_CH1_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in2_resistor_enum, ADCX140_CH2_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in3_resistor_enum, ADCX140_CH3_CFG0, 2, + resistor_text); +static SOC_ENUM_SINGLE_DECL(in4_resistor_enum, ADCX140_CH4_CFG0, 2, + resistor_text); + +static const struct snd_kcontrol_new in1_resistor_controls[] = { + SOC_DAPM_ENUM("CH1 Resistor Select", in1_resistor_enum), +}; +static const struct snd_kcontrol_new in2_resistor_controls[] = { + SOC_DAPM_ENUM("CH2 Resistor Select", in2_resistor_enum), +}; +static const struct snd_kcontrol_new in3_resistor_controls[] = { + SOC_DAPM_ENUM("CH3 Resistor Select", in3_resistor_enum), +}; +static const struct snd_kcontrol_new in4_resistor_controls[] = { + SOC_DAPM_ENUM("CH4 Resistor Select", in4_resistor_enum), +}; + +/* Analog/Digital Selection */ +static const char *adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"}; +static const char *adcx140_analog_sel_text[] = {"Analog", "Line In"}; + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1p_enum, + ADCX140_CH1_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1p_control = +SOC_DAPM_ENUM("MIC1P MUX", adcx140_mic1p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1_analog_enum, + ADCX140_CH1_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1_analog_control = +SOC_DAPM_ENUM("MIC1 Analog MUX", adcx140_mic1_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic1m_enum, + ADCX140_CH1_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic1m_control = +SOC_DAPM_ENUM("MIC1M MUX", adcx140_mic1m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2p_enum, + ADCX140_CH2_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2p_control = +SOC_DAPM_ENUM("MIC2P MUX", adcx140_mic2p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2_analog_enum, + ADCX140_CH2_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2_analog_control = +SOC_DAPM_ENUM("MIC2 Analog MUX", adcx140_mic2_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic2m_enum, + ADCX140_CH2_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic2m_control = +SOC_DAPM_ENUM("MIC2M MUX", adcx140_mic2m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3p_enum, + ADCX140_CH3_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3p_control = +SOC_DAPM_ENUM("MIC3P MUX", adcx140_mic3p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3_analog_enum, + ADCX140_CH3_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3_analog_control = +SOC_DAPM_ENUM("MIC3 Analog MUX", adcx140_mic3_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic3m_enum, + ADCX140_CH3_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic3m_control = +SOC_DAPM_ENUM("MIC3M MUX", adcx140_mic3m_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4p_enum, + ADCX140_CH4_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4p_control = +SOC_DAPM_ENUM("MIC4P MUX", adcx140_mic4p_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4_analog_enum, + ADCX140_CH4_CFG0, 7, + adcx140_analog_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4_analog_control = +SOC_DAPM_ENUM("MIC4 Analog MUX", adcx140_mic4_analog_enum); + +static SOC_ENUM_SINGLE_DECL(adcx140_mic4m_enum, + ADCX140_CH4_CFG0, 5, + adcx140_mic_sel_text); + +static const struct snd_kcontrol_new adcx140_dapm_mic4m_control = +SOC_DAPM_ENUM("MIC4M MUX", adcx140_mic4m_enum); + +static const struct snd_kcontrol_new adcx140_dapm_ch1_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 7, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 6, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + +/* Output Mixer */ +static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH2 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH3 Switch", 0, 0, 0, 0), + SOC_DAPM_SINGLE("Digital CH4 Switch", 0, 0, 0, 0), +}; + +static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { + /* Analog Differential Inputs */ + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1M"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2M"), + SND_SOC_DAPM_INPUT("MIC3P"), + SND_SOC_DAPM_INPUT("MIC3M"), + SND_SOC_DAPM_INPUT("MIC4P"), + SND_SOC_DAPM_INPUT("MIC4M"), + + SND_SOC_DAPM_OUTPUT("CH1_OUT"), + SND_SOC_DAPM_OUTPUT("CH2_OUT"), + SND_SOC_DAPM_OUTPUT("CH3_OUT"), + SND_SOC_DAPM_OUTPUT("CH4_OUT"), + SND_SOC_DAPM_OUTPUT("CH5_OUT"), + SND_SOC_DAPM_OUTPUT("CH6_OUT"), + SND_SOC_DAPM_OUTPUT("CH7_OUT"), + SND_SOC_DAPM_OUTPUT("CH8_OUT"), + + SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + &adcx140_output_mixer_controls[0], + ARRAY_SIZE(adcx140_output_mixer_controls)), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1p_control), + SND_SOC_DAPM_MUX("MIC2P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2p_control), + SND_SOC_DAPM_MUX("MIC3P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3p_control), + SND_SOC_DAPM_MUX("MIC4P Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4p_control), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1_analog_control), + SND_SOC_DAPM_MUX("MIC2 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2_analog_control), + SND_SOC_DAPM_MUX("MIC3 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3_analog_control), + SND_SOC_DAPM_MUX("MIC4 Analog Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4_analog_control), + + SND_SOC_DAPM_MUX("MIC1M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic1m_control), + SND_SOC_DAPM_MUX("MIC2M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic2m_control), + SND_SOC_DAPM_MUX("MIC3M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic3m_control), + SND_SOC_DAPM_MUX("MIC4M Input Mux", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_mic4m_control), + + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC_GAIN_CTL_CH4", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_ADC("CH1_ADC", "CH1 Capture", ADCX140_IN_CH_EN, 7, 0), + SND_SOC_DAPM_ADC("CH2_ADC", "CH2 Capture", ADCX140_IN_CH_EN, 6, 0), + SND_SOC_DAPM_ADC("CH3_ADC", "CH3 Capture", ADCX140_IN_CH_EN, 5, 0), + SND_SOC_DAPM_ADC("CH4_ADC", "CH4 Capture", ADCX140_IN_CH_EN, 4, 0), + + SND_SOC_DAPM_SWITCH("CH1_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_en_switch), + SND_SOC_DAPM_SWITCH("CH2_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_en_switch), + SND_SOC_DAPM_SWITCH("CH3_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_en_switch), + SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_en_switch), + + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_dre_en_switch), + + SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_dre_en_switch), + + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in1_resistor_controls), + SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in2_resistor_controls), + SND_SOC_DAPM_MUX("IN3 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in3_resistor_controls), + SND_SOC_DAPM_MUX("IN4 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, + in4_resistor_controls), + + SND_SOC_DAPM_MUX("Decimation Filter", SND_SOC_NOPM, 0, 0, + decimation_filter_controls), +}; + +static const struct snd_soc_dapm_route adcx140_audio_map[] = { + /* Outputs */ + {"CH1_OUT", NULL, "Output Mixer"}, + {"CH2_OUT", NULL, "Output Mixer"}, + {"CH3_OUT", NULL, "Output Mixer"}, + {"CH4_OUT", NULL, "Output Mixer"}, + + {"CH1_ASI_EN", "Switch", "CH1_ADC"}, + {"CH2_ASI_EN", "Switch", "CH2_ADC"}, + {"CH3_ASI_EN", "Switch", "CH3_ADC"}, + {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + + {"Decimation Filter", "Linear Phase", "DRE_ENABLE"}, + {"Decimation Filter", "Low Latency", "DRE_ENABLE"}, + {"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"}, + + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH4_DRE_EN"}, + + {"CH1_DRE_EN", "Switch", "CH1_ADC"}, + {"CH2_DRE_EN", "Switch", "CH2_ADC"}, + {"CH3_DRE_EN", "Switch", "CH3_ADC"}, + {"CH4_DRE_EN", "Switch", "CH4_ADC"}, + + /* Mic input */ + {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, + {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, + {"CH3_ADC", NULL, "MIC_GAIN_CTL_CH3"}, + {"CH4_ADC", NULL, "MIC_GAIN_CTL_CH4"}, + + {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH1", NULL, "IN1 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH2", NULL, "IN2 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH3", NULL, "IN3 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"}, + {"MIC_GAIN_CTL_CH4", NULL, "IN4 Analog Mic Resistor"}, + + {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1P Input Mux"}, + {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1P Input Mux"}, + {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1P Input Mux"}, + + {"IN1 Analog Mic Resistor", "2.5 kOhm", "MIC1M Input Mux"}, + {"IN1 Analog Mic Resistor", "10 kOhm", "MIC1M Input Mux"}, + {"IN1 Analog Mic Resistor", "20 kOhm", "MIC1M Input Mux"}, + + {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2P Input Mux"}, + {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2P Input Mux"}, + {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2P Input Mux"}, + + {"IN2 Analog Mic Resistor", "2.5 kOhm", "MIC2M Input Mux"}, + {"IN2 Analog Mic Resistor", "10 kOhm", "MIC2M Input Mux"}, + {"IN2 Analog Mic Resistor", "20 kOhm", "MIC2M Input Mux"}, + + {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3P Input Mux"}, + {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3P Input Mux"}, + {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3P Input Mux"}, + + {"IN3 Analog Mic Resistor", "2.5 kOhm", "MIC3M Input Mux"}, + {"IN3 Analog Mic Resistor", "10 kOhm", "MIC3M Input Mux"}, + {"IN3 Analog Mic Resistor", "20 kOhm", "MIC3M Input Mux"}, + + {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4P Input Mux"}, + {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4P Input Mux"}, + {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4P Input Mux"}, + + {"IN4 Analog Mic Resistor", "2.5 kOhm", "MIC4M Input Mux"}, + {"IN4 Analog Mic Resistor", "10 kOhm", "MIC4M Input Mux"}, + {"IN4 Analog Mic Resistor", "20 kOhm", "MIC4M Input Mux"}, + + {"MIC1 Analog Mux", "Line In", "MIC1P"}, + {"MIC2 Analog Mux", "Line In", "MIC2P"}, + {"MIC3 Analog Mux", "Line In", "MIC3P"}, + {"MIC4 Analog Mux", "Line In", "MIC4P"}, + + {"MIC1P Input Mux", "Analog", "MIC1P"}, + {"MIC1M Input Mux", "Analog", "MIC1M"}, + {"MIC2P Input Mux", "Analog", "MIC2P"}, + {"MIC2M Input Mux", "Analog", "MIC2M"}, + {"MIC3P Input Mux", "Analog", "MIC3P"}, + {"MIC3M Input Mux", "Analog", "MIC3M"}, + {"MIC4P Input Mux", "Analog", "MIC4P"}, + {"MIC4M Input Mux", "Analog", "MIC4M"}, +}; + +static const struct snd_kcontrol_new adcx140_snd_controls[] = { + SOC_SINGLE_TLV("Analog CH1 Mic Gain Volume", ADCX140_CH1_CFG1, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH2 Mic Gain Volume", ADCX140_CH1_CFG2, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH3 Mic Gain Volume", ADCX140_CH1_CFG3, 2, 42, 0, + adc_tlv), + SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, + adc_tlv), + + SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0, + dre_thresh_tlv), + SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0, + dre_gain_tlv), + + SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0, + agc_thresh_tlv), + SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0, + agc_gain_tlv), + + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH3 Out Volume", ADCX140_CH3_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH4 Out Volume", ADCX140_CH4_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH5 Out Volume", ADCX140_CH5_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH6 Out Volume", ADCX140_CH6_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH7 Out Volume", ADCX140_CH7_CFG2, + 0, 0xff, 0, dig_vol_tlv), + SOC_SINGLE_TLV("Digital CH8 Out Volume", ADCX140_CH8_CFG2, + 0, 0xff, 0, dig_vol_tlv), +}; + +static int adcx140_reset(struct adcx140_priv *adcx140) +{ + int ret = 0; + + if (adcx140->gpio_reset) { + gpiod_direction_output(adcx140->gpio_reset, 0); + /* 8.4.1: wait for hw shutdown (25ms) + >= 1ms */ + usleep_range(30000, 100000); + gpiod_direction_output(adcx140->gpio_reset, 1); + } else { + ret = regmap_write(adcx140->regmap, ADCX140_SW_RESET, + ADCX140_RESET); + } + + /* 8.4.2: wait >= 10 ms after entering sleep mode. */ + usleep_range(10000, 100000); + + return 0; +} + +static int adcx140_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + u8 data = 0; + + switch (params_width(params)) { + case 16: + data = ADCX140_16_BIT_WORD; + break; + case 20: + data = ADCX140_20_BIT_WORD; + break; + case 24: + data = ADCX140_24_BIT_WORD; + break; + case 32: + data = ADCX140_32_BIT_WORD; + break; + default: + dev_err(component->dev, "%s: Unsupported width %d\n", + __func__, params_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, + ADCX140_WORD_LEN_MSK, data); + + return 0; +} + +static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + u8 iface_reg1 = 0; + u8 iface_reg2 = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg2 |= ADCX140_BCLK_FSYNC_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(component->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* signal polarity */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_IF: + iface_reg1 |= ADCX140_FSYNCINV_BIT; + break; + case SND_SOC_DAIFMT_IB_IF: + iface_reg1 |= ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT; + break; + case SND_SOC_DAIFMT_IB_NF: + iface_reg1 |= ADCX140_BCLKINV_BIT; + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + dev_err(component->dev, "Invalid DAI clock signal polarity\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg1 |= ADCX140_I2S_MODE_BIT; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= ADCX140_LEFT_JUST_BIT; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + break; + default: + dev_err(component->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, + ADCX140_FSYNCINV_BIT | + ADCX140_BCLKINV_BIT | + ADCX140_ASI_FORMAT_MSK, + iface_reg1); + snd_soc_component_update_bits(component, ADCX140_MST_CFG0, + ADCX140_BCLK_FSYNC_MASTER, iface_reg2); + + return 0; +} + +static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = codec_dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + unsigned int lsb; + + if (tx_mask != rx_mask) { + dev_err(component->dev, "tx and rx masks must be symmetric\n"); + return -EINVAL; + } + + /* TDM based on DSP mode requires slots to be adjacent */ + lsb = __ffs(tx_mask); + if ((lsb + 1) != __fls(tx_mask)) { + dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + return -EINVAL; + } + + switch (slot_width) { + case 16: + case 20: + case 24: + case 32: + break; + default: + dev_err(component->dev, "Unsupported slot width %d\n", slot_width); + return -EINVAL; + } + + adcx140->tdm_delay = lsb; + adcx140->slot_width = slot_width; + + return 0; +} + +static int adcx140_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int offset = 0; + int width = adcx140->slot_width; + + if (!width) + width = substream->runtime->sample_bits; + + /* TDM slot selection only valid in DSP_A/_B mode */ + if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_A) + offset += (adcx140->tdm_delay * width + 1); + else if (adcx140->dai_fmt == SND_SOC_DAIFMT_DSP_B) + offset += adcx140->tdm_delay * width; + + /* Configure data offset */ + snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, + ADCX140_TX_OFFSET_MASK, offset); + + return 0; +} + +static const struct snd_soc_dai_ops adcx140_dai_ops = { + .hw_params = adcx140_hw_params, + .set_fmt = adcx140_set_dai_fmt, + .prepare = adcx140_prepare, + .set_tdm_slot = adcx140_set_dai_tdm_slot, +}; + +static int adcx140_codec_probe(struct snd_soc_component *component) +{ + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int sleep_cfg_val = ADCX140_WAKE_DEV; + u8 bias_source; + u8 vref_source; + int ret; + + ret = device_property_read_u8(adcx140->dev, "ti,mic-bias-source", + &bias_source); + if (ret) + bias_source = ADCX140_MIC_BIAS_VAL_VREF; + + if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || + bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { + dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + return -EINVAL; + } + + ret = device_property_read_u8(adcx140->dev, "ti,vref-source", + &vref_source); + if (ret) + vref_source = ADCX140_MIC_BIAS_VREF_275V; + + if (vref_source < ADCX140_MIC_BIAS_VREF_275V || + vref_source > ADCX140_MIC_BIAS_VREF_1375V) { + dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + return -EINVAL; + } + + bias_source |= vref_source; + + ret = adcx140_reset(adcx140); + if (ret) + goto out; + + if(adcx140->supply_areg == NULL) + sleep_cfg_val |= ADCX140_AREG_INTERNAL; + + ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val); + if (ret) { + dev_err(adcx140->dev, "setting sleep config failed %d\n", ret); + goto out; + } + + /* 8.4.3: Wait >= 1ms after entering active mode. */ + usleep_range(1000, 100000); + + ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG, + ADCX140_MIC_BIAS_VAL_MSK | + ADCX140_MIC_BIAS_VREF_MSK, bias_source); + if (ret) + dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); +out: + return ret; +} + +static int adcx140_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); + int pwr_cfg = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | + ADCX140_PWR_CFG_ADC_PDZ; + break; + case SND_SOC_BIAS_OFF: + pwr_cfg = 0x0; + break; + } + + return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); +} + +static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { + .probe = adcx140_codec_probe, + .set_bias_level = adcx140_set_bias_level, + .controls = adcx140_snd_controls, + .num_controls = ARRAY_SIZE(adcx140_snd_controls), + .dapm_widgets = adcx140_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adcx140_dapm_widgets), + .dapm_routes = adcx140_audio_map, + .num_dapm_routes = ARRAY_SIZE(adcx140_audio_map), + .suspend_bias_off = 1, + .idle_bias_on = 0, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver adcx140_dai_driver[] = { + { + .name = "tlv320adcx140-codec", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = ADCX140_MAX_CHANNELS, + .rates = ADCX140_RATES, + .formats = ADCX140_FORMATS, + }, + .ops = &adcx140_dai_ops, + .symmetric_rates = 1, + } +}; + +static const struct of_device_id tlv320adcx140_of_match[] = { + { .compatible = "ti,tlv320adc3140" }, + { .compatible = "ti,tlv320adc5140" }, + { .compatible = "ti,tlv320adc6140" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adcx140_of_match); + +static int adcx140_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct adcx140_priv *adcx140; + int ret; + + adcx140 = devm_kzalloc(&i2c->dev, sizeof(*adcx140), GFP_KERNEL); + if (!adcx140) + return -ENOMEM; + + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(adcx140->gpio_reset)) + dev_info(&i2c->dev, "Reset GPIO not defined\n"); + + adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev, + "areg"); + if (IS_ERR(adcx140->supply_areg)) { + if (PTR_ERR(adcx140->supply_areg) == -EPROBE_DEFER) + return -EPROBE_DEFER; + else + adcx140->supply_areg = NULL; + } else { + ret = regulator_enable(adcx140->supply_areg); + if (ret) { + dev_err(adcx140->dev, "Failed to enable areg\n"); + return ret; + } + } + + adcx140->regmap = devm_regmap_init_i2c(i2c, &adcx140_i2c_regmap); + if (IS_ERR(adcx140->regmap)) { + ret = PTR_ERR(adcx140->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); + + return devm_snd_soc_register_component(&i2c->dev, + &soc_codec_driver_adcx140, + adcx140_dai_driver, 1); +} + +static const struct i2c_device_id adcx140_i2c_id[] = { + { "tlv320adc3140", 0 }, + { "tlv320adc5140", 1 }, + { "tlv320adc6140", 2 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adcx140_i2c_id); + +static struct i2c_driver adcx140_i2c_driver = { + .driver = { + .name = "tlv320adcx140-codec", + .of_match_table = of_match_ptr(tlv320adcx140_of_match), + }, + .probe = adcx140_i2c_probe, + .id_table = adcx140_i2c_id, +}; +module_i2c_driver(adcx140_i2c_driver); + +MODULE_AUTHOR("Dan Murphy <dmurphy@ti.com>"); +MODULE_DESCRIPTION("ASoC TLV320ADCX140 CODEC Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h new file mode 100644 index 000000000000..6d055e55909e --- /dev/null +++ b/sound/soc/codecs/tlv320adcx140.h @@ -0,0 +1,131 @@ +// SPDX-License-Identifier: GPL-2.0 +// TLV320ADCX104 Sound driver +// Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com/ + +#ifndef _TLV320ADCX140_H +#define _TLV320ADCX140_H + +#define ADCX140_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define ADCX140_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define ADCX140_PAGE_SELECT 0x00 +#define ADCX140_SW_RESET 0x01 +#define ADCX140_SLEEP_CFG 0x02 +#define ADCX140_SHDN_CFG 0x05 +#define ADCX140_ASI_CFG0 0x07 +#define ADCX140_ASI_CFG1 0x08 +#define ADCX140_ASI_CFG2 0x09 +#define ADCX140_ASI_CH1 0x0b +#define ADCX140_ASI_CH2 0x0c +#define ADCX140_ASI_CH3 0x0d +#define ADCX140_ASI_CH4 0x0e +#define ADCX140_ASI_CH5 0x0f +#define ADCX140_ASI_CH6 0x10 +#define ADCX140_ASI_CH7 0x11 +#define ADCX140_ASI_CH8 0x12 +#define ADCX140_MST_CFG0 0x13 +#define ADCX140_MST_CFG1 0x14 +#define ADCX140_ASI_STS 0x15 +#define ADCX140_CLK_SRC 0x16 +#define ADCX140_PDMCLK_CFG 0x1f +#define ADCX140_PDM_CFG 0x20 +#define ADCX140_GPIO_CFG0 0x21 +#define ADCX140_GPO_CFG1 0x22 +#define ADCX140_GPO_CFG2 0x23 +#define ADCX140_GPO_CFG3 0x24 +#define ADCX140_GPO_CFG4 0x25 +#define ADCX140_GPO_VAL 0x29 +#define ADCX140_GPIO_MON 0x2a +#define ADCX140_GPI_CFG0 0x2b +#define ADCX140_GPI_CFG1 0x2c +#define ADCX140_GPI_MON 0x2f +#define ADCX140_INT_CFG 0x32 +#define ADCX140_INT_MASK0 0x33 +#define ADCX140_INT_LTCH0 0x36 +#define ADCX140_BIAS_CFG 0x3b +#define ADCX140_CH1_CFG0 0x3c +#define ADCX140_CH1_CFG1 0x3d +#define ADCX140_CH1_CFG2 0x3e +#define ADCX140_CH1_CFG3 0x3f +#define ADCX140_CH1_CFG4 0x40 +#define ADCX140_CH2_CFG0 0x41 +#define ADCX140_CH2_CFG1 0x42 +#define ADCX140_CH2_CFG2 0x43 +#define ADCX140_CH2_CFG3 0x44 +#define ADCX140_CH2_CFG4 0x45 +#define ADCX140_CH3_CFG0 0x46 +#define ADCX140_CH3_CFG1 0x47 +#define ADCX140_CH3_CFG2 0x48 +#define ADCX140_CH3_CFG3 0x49 +#define ADCX140_CH3_CFG4 0x4a +#define ADCX140_CH4_CFG0 0x4b +#define ADCX140_CH4_CFG1 0x4c +#define ADCX140_CH4_CFG2 0x4d +#define ADCX140_CH4_CFG3 0x4e +#define ADCX140_CH4_CFG4 0x4f +#define ADCX140_CH5_CFG2 0x52 +#define ADCX140_CH5_CFG3 0x53 +#define ADCX140_CH5_CFG4 0x54 +#define ADCX140_CH6_CFG2 0x57 +#define ADCX140_CH6_CFG3 0x58 +#define ADCX140_CH6_CFG4 0x59 +#define ADCX140_CH7_CFG2 0x5c +#define ADCX140_CH7_CFG3 0x5d +#define ADCX140_CH7_CFG4 0x5e +#define ADCX140_CH8_CFG2 0x61 +#define ADCX140_CH8_CFG3 0x62 +#define ADCX140_CH8_CFG4 0x63 +#define ADCX140_DSP_CFG0 0x6b +#define ADCX140_DSP_CFG1 0x6c +#define ADCX140_DRE_CFG0 0x6d +#define ADCX140_AGC_CFG0 0x70 +#define ADCX140_IN_CH_EN 0x73 +#define ADCX140_ASI_OUT_CH_EN 0x74 +#define ADCX140_PWR_CFG 0x75 +#define ADCX140_DEV_STS0 0x76 +#define ADCX140_DEV_STS1 0x77 + +#define ADCX140_RESET BIT(0) + +#define ADCX140_WAKE_DEV BIT(0) +#define ADCX140_AREG_INTERNAL BIT(7) + +#define ADCX140_BCLKINV_BIT BIT(2) +#define ADCX140_FSYNCINV_BIT BIT(3) +#define ADCX140_INV_MSK (ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT) +#define ADCX140_BCLK_FSYNC_MASTER BIT(7) +#define ADCX140_I2S_MODE_BIT BIT(6) +#define ADCX140_LEFT_JUST_BIT BIT(7) +#define ADCX140_ASI_FORMAT_MSK (ADCX140_I2S_MODE_BIT | ADCX140_LEFT_JUST_BIT) + +#define ADCX140_16_BIT_WORD 0x0 +#define ADCX140_20_BIT_WORD BIT(4) +#define ADCX140_24_BIT_WORD BIT(5) +#define ADCX140_32_BIT_WORD (BIT(4) | BIT(5)) +#define ADCX140_WORD_LEN_MSK 0x30 + +#define ADCX140_MAX_CHANNELS 8 + +#define ADCX140_MIC_BIAS_VAL_VREF 0 +#define ADCX140_MIC_BIAS_VAL_VREF_1096 1 +#define ADCX140_MIC_BIAS_VAL_AVDD 6 +#define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4) + +#define ADCX140_MIC_BIAS_VREF_275V 0 +#define ADCX140_MIC_BIAS_VREF_25V 1 +#define ADCX140_MIC_BIAS_VREF_1375V 2 +#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) + +#define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) +#define ADCX140_PWR_CFG_ADC_PDZ BIT(6) +#define ADCX140_PWR_CFG_PLL_PDZ BIT(5) + +#define ADCX140_TX_OFFSET_MASK GENMASK(4, 0) + +#endif /* _TLV320ADCX140_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f11ffa28683b..700cc1212770 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4926,11 +4926,11 @@ static const struct regmap_range_cfg wcd9335_ranges[] = { .name = "WCD9335", .range_min = 0x0, .range_max = WCD9335_MAX_REGISTER, - .selector_reg = WCD9335_REG(0x0, 0), + .selector_reg = WCD9335_SEL_REGISTER, .selector_mask = 0xff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x100, }, }; @@ -4968,12 +4968,12 @@ static const struct regmap_range_cfg wcd9335_ifc_ranges[] = { { .name = "WCD9335-IFC-DEV", .range_min = 0x0, - .range_max = WCD9335_REG(0, 0x7ff), - .selector_reg = WCD9335_REG(0, 0x0), - .selector_mask = 0xff, + .range_max = WCD9335_MAX_REGISTER, + .selector_reg = WCD9335_SEL_REGISTER, + .selector_mask = 0xfff, .selector_shift = 0, - .window_start = 0x0, - .window_len = 0x1000, + .window_start = 0x800, + .window_len = 0x400, }, }; @@ -4981,7 +4981,7 @@ static struct regmap_config wcd9335_ifc_regmap_config = { .reg_bits = 16, .val_bits = 8, .can_multi_write = true, - .max_register = WCD9335_REG(0, 0x7FF), + .max_register = WCD9335_MAX_REGISTER, .ranges = wcd9335_ifc_ranges, .num_ranges = ARRAY_SIZE(wcd9335_ifc_ranges), }; diff --git a/sound/soc/codecs/wcd9335.h b/sound/soc/codecs/wcd9335.h index 4d9be2496c30..72060824c743 100644 --- a/sound/soc/codecs/wcd9335.h +++ b/sound/soc/codecs/wcd9335.h @@ -8,9 +8,9 @@ * in slimbus mode the reg base starts from 0x800 * in i2s/i2c mode the reg base is 0x0 */ -#define WCD9335_REG(pg, r) ((pg << 12) | (r) | 0x800) +#define WCD9335_REG(pg, r) ((pg << 8) | (r)) #define WCD9335_REG_OFFSET(r) (r & 0xFF) -#define WCD9335_PAGE_OFFSET(r) ((r >> 12) & 0xFF) +#define WCD9335_PAGE_OFFSET(r) ((r >> 8) & 0xFF) /* Page-0 Registers */ #define WCD9335_PAGE0_PAGE_REGISTER WCD9335_REG(0x00, 0x000) @@ -600,7 +600,8 @@ #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE BIT(0) #define WCD9335_CDC_CLK_RST_CTRL_FS_CNT_DISABLE 0 #define WCD9335_CDC_TOP_TOP_CFG1 WCD9335_REG(0x0d, 0x082) -#define WCD9335_MAX_REGISTER WCD9335_REG(0x80, 0x0FF) +#define WCD9335_MAX_REGISTER 0xffff +#define WCD9335_SEL_REGISTER 0x800 /* SLIMBUS Slave Registers */ #define WCD9335_SLIM_PGD_PORT_INT_EN0 WCD9335_REG(0, 0x30) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 158e878abd6c..5269857e2746 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -3,7 +3,6 @@ #include <linux/clk.h> #include <linux/clk-provider.h> -#include <linux/gpio.h> #include <linux/interrupt.h> #include <linux/kernel.h> #include <linux/mfd/wcd934x/registers.h> @@ -11,10 +10,7 @@ #include <linux/module.h> #include <linux/mutex.h> #include <linux/of_clk.h> -#include <linux/of_device.h> -#include <linux/of_gpio.h> #include <linux/of.h> -#include <linux/of_irq.h> #include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -1202,11 +1198,6 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, WCD934X_ANA_RCO_BG_EN_MASK, 0); usleep_range(100, 110); - } else if (sido_src == SIDO_SOURCE_RCO_BG) { - regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, - WCD934X_ANA_RCO_BG_EN_MASK, - WCD934X_ANA_RCO_BG_ENABLE); - usleep_range(100, 110); regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, WCD934X_ANA_BUCK_PRE_EN1_MASK, WCD934X_ANA_BUCK_PRE_EN1_ENABLE); @@ -1219,6 +1210,11 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) WCD934X_ANA_BUCK_HI_ACCU_EN_MASK, WCD934X_ANA_BUCK_HI_ACCU_ENABLE); usleep_range(100, 110); + } else if (sido_src == SIDO_SOURCE_RCO_BG) { + regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, + WCD934X_ANA_RCO_BG_EN_MASK, + WCD934X_ANA_RCO_BG_ENABLE); + usleep_range(100, 110); } wcd->sido_input_src = sido_src; @@ -1883,20 +1879,16 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; @@ -3392,18 +3384,15 @@ static void wcd934x_codec_hphdelay_lutbypass(struct snd_soc_component *comp, { u8 hph_dly_mask; u16 hph_lut_bypass_reg = 0; - u16 hph_comp_ctrl7 = 0; switch (interp_idx) { case INTERP_HPHL: hph_dly_mask = 1; hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHL_COMP_LUT; - hph_comp_ctrl7 = WCD934X_CDC_COMPANDER1_CTL7; break; case INTERP_HPHR: hph_dly_mask = 2; hph_lut_bypass_reg = WCD934X_CDC_TOP_HPHR_COMP_LUT; - hph_comp_ctrl7 = WCD934X_CDC_COMPANDER2_CTL7; break; default: return; diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 727d6703c905..fbcee21736e8 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -43,7 +43,7 @@ struct dfw_binrec { u8 command; u32 length:24; u32 address; - uint8_t data[0]; + uint8_t data[]; } __packed; struct dfw_inforec { diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9dc215b5c504..499e87d1dfcc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2245,14 +2245,14 @@ static int wm5110_open(struct snd_compr_stream *stream) struct arizona *arizona = priv->core.arizona; int n_adsp; - if (strcmp(rtd->codec_dai->name, "wm5110-dsp-voicectrl") == 0) { + if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-voicectrl") == 0) { n_adsp = 2; - } else if (strcmp(rtd->codec_dai->name, "wm5110-dsp-trace") == 0) { + } else if (strcmp(asoc_rtd_to_codec(rtd, 0)->name, "wm5110-dsp-trace") == 0) { n_adsp = 0; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); return -EINVAL; } diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index dc4fe4f5239d..06ba36595ddd 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -196,14 +196,6 @@ SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0), SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0), }; -/* AUX Input boost vol */ -static const struct snd_kcontrol_new wm8974_aux_boost_controls = -SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0); - -/* Mic Input boost vol */ -static const struct snd_kcontrol_new wm8974_mic_boost_controls = -SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0); - static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0, &wm8974_speaker_mixer_controls[0], diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3d32b501aca..1ef69409ccd1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1436,12 +1436,12 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, subname = NULL; /* don't append subname */ break; case 2: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s%c %.12s %x", dsp->name, *region_name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; default: - ret = snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %.12s %x", dsp->name, wm_adsp_fw_text[dsp->fw], alg_region->alg); break; @@ -3467,22 +3467,22 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) if (wm_adsp_fw[dsp->fw].num_caps == 0) { adsp_err(dsp, "%s: Firmware does not support compressed API\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -ENXIO; goto out; } if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) { adsp_err(dsp, "%s: Firmware does not support stream direction\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -EINVAL; goto out; } list_for_each_entry(tmp, &dsp->compr_list, list) { - if (!strcmp(tmp->name, rtd->codec_dai->name)) { + if (!strcmp(tmp->name, asoc_rtd_to_codec(rtd, 0)->name)) { adsp_err(dsp, "%s: Only a single stream supported per dai\n", - rtd->codec_dai->name); + asoc_rtd_to_codec(rtd, 0)->name); ret = -EBUSY; goto out; } @@ -3496,7 +3496,7 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) compr->dsp = dsp; compr->stream = stream; - compr->name = rtd->codec_dai->name; + compr->name = asoc_rtd_to_codec(rtd, 0)->name; list_add_tail(&compr->list, &dsp->compr_list); diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index b59f1d0e7f84..f2d6f2f81f14 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -676,7 +676,6 @@ struct wsa881x_priv { int active_ports; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; - bool stream_prepared; }; static void wsa881x_init(struct wsa881x_priv *wsa881x) @@ -954,41 +953,6 @@ static const struct snd_soc_dapm_widget wsa881x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR"), }; -static int wsa881x_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - int ret; - - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - wsa881x->stream_prepared = false; - } - - - ret = sdw_prepare_stream(wsa881x->sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual PA enable. PA enable should only happen after - * soundwire ports are enabled if not DC on the line is accumulated - * resulting in Click/Pop Noise - * PA enable/mute are handled as part of DAPM and digital mute. - */ - - ret = sdw_enable_stream(wsa881x->sruntime); - if (ret) { - sdw_deprepare_stream(wsa881x->sruntime); - return ret; - } - wsa881x->stream_prepared = true; - - return ret; -} - static int wsa881x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1016,12 +980,7 @@ static int wsa881x_hw_free(struct snd_pcm_substream *substream, { struct wsa881x_priv *wsa881x = dev_get_drvdata(dai->dev); - if (wsa881x->stream_prepared) { - sdw_disable_stream(wsa881x->sruntime); - sdw_deprepare_stream(wsa881x->sruntime); - sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); - wsa881x->stream_prepared = false; - } + sdw_stream_remove_slave(wsa881x->slave, wsa881x->sruntime); return 0; } @@ -1052,7 +1011,6 @@ static int wsa881x_digital_mute(struct snd_soc_dai *dai, int mute, int stream) static struct snd_soc_dai_ops wsa881x_dai_ops = { .hw_params = wsa881x_hw_params, - .prepare = wsa881x_prepare, .hw_free = wsa881x_hw_free, .mute_stream = wsa881x_digital_mute, .set_sdw_stream = wsa881x_set_sdw_stream, @@ -1150,7 +1108,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, wsa881x->sconfig.type = SDW_STREAM_PDM; pdev->prop.sink_ports = GENMASK(WSA881X_MAX_SWR_PORTS, 0); pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; - gpiod_set_value(wsa881x->sd_n, 1); + gpiod_direction_output(wsa881x->sd_n, 1); wsa881x->regmap = devm_regmap_init_sdw(pdev, &wsa881x_regmap_config); if (IS_ERR(wsa881x->regmap)) { diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7eeca2150b2d..515f88456dbd 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component) { struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component); struct snd_soc_dai *dai; + int stream; if (dev->capability & DW_I2S_MASTER) clk_enable(dev->clk); for_each_component_dais(component, dai) { - if (dai->playback_active) - dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK); - if (dai->capture_active) - dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE); + for_each_pcm_streams(stream) + if (dai->stream_active[stream]) + dw_i2s_config(dev, stream); } return 0; diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index 4b25aca3804f..9868e7373d36 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -140,7 +140,7 @@ static int dw_pcm_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); snd_soc_set_runtime_hwparams(substream, &dw_pcm_hardware); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 6f3b768489f6..4ff2d21bb32f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -31,8 +31,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9ce55feaac22..bb33601fab84 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -159,7 +159,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return 0; /* Specific configurations of DAIs starts from here */ - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx], + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx], cpu_priv->sysclk_freq[tx], cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { @@ -168,7 +168,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, } if (cpu_priv->slot_width) { - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); @@ -257,7 +257,7 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -446,14 +446,14 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); struct snd_soc_pcm_runtime *rtd = list_first_entry( &card->rtd_list, struct snd_soc_pcm_runtime, list); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; int ret; if (fsl_asoc_card_is_ac97(priv)) { #if IS_ENABLED(CONFIG_SND_AC97_CODEC) - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component); /* diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ece130f59d15..e7178817d7a7 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -152,7 +152,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; - struct snd_soc_dai *dai = be->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(be, 0); if (dpcm->fe != rtd) continue; @@ -169,7 +169,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, } /* Override dma_data of the Front-End and config its dmaengine */ - dma_params_fe = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params_fe = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_params_fe->addr = asrc_priv->paddr + REG_ASRDx(!dir, index); dma_params_fe->maxburst = dma_params_be->maxburst; @@ -328,7 +328,7 @@ static int fsl_asrc_dma_startup(struct snd_soc_component *component, goto dma_chan_err; } - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); /* Refine the snd_imx_hardware according to caps of DMA. */ ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component, return ret; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 7858a5499ac5..c711d2d93280 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -370,7 +370,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, int sample_rate) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; struct regmap *regmap = spdif_priv->regmap; struct platform_device *pdev = spdif_priv->pdev; @@ -458,7 +458,7 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; u32 scr, mask; @@ -534,7 +534,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; u32 scr, mask, i; @@ -569,7 +569,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; struct platform_device *pdev = spdif_priv->pdev; u32 sample_rate = params_rate(params); @@ -597,7 +597,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 intr = SIE_INTR_FOR(tx); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5c97269be346..bad89b0d129e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -631,7 +631,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); int ret; ret = clk_prepare_enable(ssi->clk); @@ -655,7 +655,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); clk_disable_unprepare(ssi->clk); } @@ -854,7 +854,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); if (fsl_ssi_is_i2s_master(ssi) && ssi->baudclk_streams & BIT(substream->stream)) { @@ -1059,7 +1059,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; switch (cmd) { diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 5ef6881395e0..e09b45de0efd 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -85,13 +85,13 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; /* set DAI configuration */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret) { dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); return ret; } - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), FSL_SAI_CLK_MAST1, 0, dir); if (ret) { dev_err(dev, "failed to set cpu sysclk: %d\n", ret); return ret; @@ -101,7 +101,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, * Per datasheet, AUDMIX expects 8 slots and 32 bits * for every slot in TDM mode. */ - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), BIT(channels) - 1, BIT(channels) - 1, 8, 32); if (ret) dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); @@ -125,7 +125,7 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, fmt |= SND_SOC_DAIFMT_CBM_CFM; /* set AUDMIX DAI configuration */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret) dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 2b679680c93f..fab2d6c56653 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -27,8 +27,8 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 15e8b9343c35..f45cb4bbb6c4 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -30,7 +30,7 @@ static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) struct device *dev = rtd->card->dev; int ret; - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), SGTL5000_SYSCLK, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "could not set codec driver clock params\n"); diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index ed7211d744b3..3b8c796d7829 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -115,7 +115,7 @@ static int psc_dma_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct snd_pcm_runtime *runtime = substream->runtime; struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; @@ -217,7 +217,7 @@ static int psc_dma_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; int rc; @@ -245,7 +245,7 @@ static int psc_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream); @@ -271,7 +271,7 @@ psc_dma_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct psc_dma_stream *s; dma_addr_t count; @@ -298,7 +298,7 @@ static int psc_dma_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; size_t size = psc_dma_hardware.buffer_bytes_max; int rc; diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9bc01f374b39..1ab4fbda08cb 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -39,7 +39,7 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); u32 mode; dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 23617eb09ba1..f7bd90051ce7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -105,7 +105,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, machine_data->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), machine_data->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format\n"); return ret; @@ -115,7 +115,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, machine_data->clk_frequency, machine_data->codec_clk_direction); if (ret < 0) { diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index 38ac4a397742..a36d4e8cd55c 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -37,8 +37,8 @@ static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 6114b01b90f7..fe3091590f20 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -128,7 +128,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format\n"); return ret; @@ -138,7 +138,7 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), 0, mdata->clk_frequency, mdata->codec_clk_direction); if (ret < 0) { dev_err(dev, "could not set codec driver clock params\n"); diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index 72687235c0ae..f5374fe354ab 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -134,14 +134,14 @@ static int p1022_rdk_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format (ret=%i)\n", ret); return ret; } - ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency, + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, mdata->clk_frequency, mdata->clk_frequency); if (ret < 0) { dev_err(dev, "could not set codec PLL frequency (ret=%i)\n", diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 52d321bede9c..8b1551c55452 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -76,8 +76,8 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int i, found = 0; snd_pcm_format_t format = params_format(params); unsigned int rate = params_rate(params); @@ -196,7 +196,7 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Headphone jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 9b794775df53..8c54dc6710fe 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,8 +213,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -249,8 +249,8 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -331,22 +331,70 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } +static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, + struct simple_dai_props *dai_props) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_component *component; + struct snd_soc_pcm_stream *params; + struct snd_pcm_hardware hw; + int i, ret, stream; + + /* Only codecs should have non_legacy_dai_naming set. */ + for_each_rtd_components(rtd, i, component) { + if (!component->driver->non_legacy_dai_naming) + return 0; + } + + /* Assumes the capabilities are the same for all supported streams */ + for_each_pcm_streams(stream) { + ret = snd_soc_runtime_calc_hw(rtd, &hw, stream); + if (ret == 0) + break; + } + + if (ret < 0) { + dev_err(rtd->dev, "simple-card: no valid dai_link params\n"); + return ret; + } + + params = devm_kzalloc(rtd->dev, sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + + params->formats = hw.formats; + params->rates = hw.rates; + params->rate_min = hw.rate_min; + params->rate_max = hw.rate_max; + params->channels_min = hw.channels_min; + params->channels_max = hw.channels_max; + + dai_link->params = params; + dai_link->num_params = 1; + + return 0; +} + int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) { struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_init_dai(rtd->codec_dai, + ret = asoc_simple_init_dai(asoc_rtd_to_codec(rtd, 0), dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_init_dai(rtd->cpu_dai, + ret = asoc_simple_init_dai(asoc_rtd_to_cpu(rtd, 0), dai_props->cpu_dai); if (ret < 0) return ret; + ret = asoc_simple_init_dai_link_params(rtd, dai_props); + if (ret < 0) + return ret; + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index fdd2c73fd2fa..a495d1050d49 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -397,7 +397,7 @@ static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_dmaengine_dai_dma_data *dma_data; int ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st); ret = snd_hwparams_to_dma_slave_config(st, params, sc); if (ret) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 4b1853409633..db052ec17d5d 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -403,7 +403,7 @@ static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_dmaengine_dai_dma_data *dma_data; int ret; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, st); + dma_data = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), st); ret = snd_hwparams_to_dma_slave_config(st, params, sc); if (ret) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f1..f883c9340eee 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 340bd2be39a7..82f2b6357778 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -649,7 +649,7 @@ static snd_pcm_uframes_t sst_soc_pointer(struct snd_soc_component *component, static int sst_soc_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; if (dai->driver->playback.channels_min || @@ -741,7 +741,7 @@ static int sst_soc_prepare(struct device *dev) /* set the SSPs to idle */ for_each_card_rtds(drv->soc_card, rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); if (dai->active) { send_ssp_cmd(dai, dai->name, 0); @@ -762,7 +762,7 @@ static void sst_soc_complete(struct device *dev) /* restart SSPs */ for_each_card_rtds(drv->soc_card, rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); if (dai->active) { sst_handle_vb_timer(dai, true); diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c index d952719bc098..5862fe968083 100644 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ b/sound/soc/intel/atom/sst/sst_pci.c @@ -99,7 +99,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); do_release_regions: pci_release_regions(pci); - return 0; + return ret; } /* diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 9ca2567d0059..556c3104e641 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -289,7 +289,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON @@ -302,6 +301,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms @@ -402,6 +402,7 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH tristate "GLK with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Geminilake platforms @@ -413,10 +414,10 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT5682 select SND_SOC_MAX98357A select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Geminilake platforms @@ -430,7 +431,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "SKL/KBL/BXT/APL with HDA Codecs" - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC + depends on SND_HDA_CODEC_HDMI select SND_SOC_HDAC_HDMI select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected @@ -448,15 +449,31 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on I2C && ACPI depends on (SND_SOC_SOF_HDA_LINK && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI + select SND_SOC_MAX98373 + select SND_SOC_RT1015 select SND_SOC_RT5682 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms with rt5682 codec. Say Y if you have such a device. If unsure select "N". + +config SND_SOC_INTEL_SOF_PCM512x_MACH + tristate "SOF with TI PCM512x codec" + depends on I2C && ACPI + depends on (SND_SOC_SOF_HDA_AUDIO_CODEC && (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ + (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + depends on SND_HDA_CODEC_HDMI + select SND_SOC_PCM512x_I2C + help + This adds support for ASoC machine driver for SOF platforms + with TI PCM512x I2S audio codec. + Say Y or m if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK) @@ -476,11 +493,11 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH tristate "CML with RT1011 and RT5682 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_RT1011 select SND_SOC_RT5682 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platform with RT1011 + RT5682 I2S codec. @@ -492,19 +509,43 @@ endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK if SND_SOC_SOF_JASPERLAKE config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH - tristate "SOF with DA7219 and MAX98373 in I2S Mode" + tristate "SOF with DA7219 and MAX98373/MAX98360A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI select SND_SOC_DA7219 select SND_SOC_MAX98373 select SND_SOC_DMIC - select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC help This adds support for ASoC machine driver for SOF platforms - with DA7219 + MAX98373 I2S audio codec. + with DA7219 + MAX98373/MAX98360A I2S audio codec. Say Y if you have such a device. If unsure select "N". endif ## SND_SOC_SOF_JASPERLAKE +if SND_SOC_SOF_INTEL_SOUNDWIRE + +config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH + tristate "SoundWire generic machine driver" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST + depends on SOUNDWIRE + depends on SND_HDA_CODEC_HDMI + select SND_SOC_RT700_SDW + select SND_SOC_RT711_SDW + select SND_SOC_RT1308_SDW + select SND_SOC_RT1308 + select SND_SOC_RT715_SDW + select SND_SOC_RT5682_SDW + select SND_SOC_DMIC + help + Add support for Intel SoundWire-based platforms connected to + RT700, RT711, RT1308 and RT715 + If unsure select "N". + +endif + + endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index b74ddd49bd39..1ef6e60bc2a0 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -7,6 +7,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o hda_dsp_common.o +snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o hda_dsp_common.o snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o hda_dsp_common.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o @@ -18,7 +19,7 @@ snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o +snd-soc-sof_rt5682-objs := sof_rt5682.o hda_dsp_common.o sof_maxim_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o hda_dsp_common.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -30,13 +31,18 @@ snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_c snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o - +snd-soc-sof-sdw-objs += sof_sdw.o \ + sof_sdw_rt711.o sof_sdw_rt700.o \ + sof_sdw_rt1308.o sof_sdw_rt715.o \ + sof_sdw_rt5682.o \ + sof_sdw_dmic.o sof_sdw_hdmi.o hda_dsp_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o @@ -62,4 +68,4 @@ obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max9 obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o - +obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 1a302436d450..6c2fdb5659ed 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -107,7 +107,7 @@ static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* Workaround: set codec PLL to 19.2MHz that PLL source is @@ -166,8 +166,8 @@ static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) { struct bdw_rt5650_priv *bdw_rt5650 = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; int ret; /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. @@ -226,9 +226,6 @@ SND_SOC_DAILINK_DEF(be, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif static struct snd_soc_dai_link bdw_rt5650_dais[] = { @@ -264,7 +261,11 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, .init = bdw_rt5650_init, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, be, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, be, platform), +#endif }, }; @@ -298,7 +299,7 @@ static int bdw_rt5650_probe(struct platform_device *pdev) return -ENOMEM; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card, mach->mach_params.platform); diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index bb643c99069d..6b4b64098d36 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -157,7 +157,7 @@ static int bdw_rt5677_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, 24576000, @@ -174,7 +174,7 @@ static int bdw_rt5677_dsp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_PLL1, 24576000, @@ -226,7 +226,7 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) { struct bdw_rt5677_priv *bdw_rt5677 = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); int ret; @@ -298,9 +298,6 @@ SND_SOC_DAILINK_DEF(be, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif /* Wake on voice interface */ @@ -350,7 +347,11 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, .init = bdw_rt5677_init, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, be, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, be, platform), +#endif }, }; @@ -412,7 +413,7 @@ static int bdw_rt5677_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5677_card, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index b9c12e24c70b..acb4e36682cb 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -70,7 +70,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, @@ -104,7 +104,7 @@ static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, @@ -167,9 +167,6 @@ SND_SOC_DAILINK_DEF(codec, #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); -#else -SND_SOC_DAILINK_DEF(ssp0_port, - DAILINK_COMP_ARRAY(COMP_DUMMY())); #endif /* broadwell digital audio interface glue - connects codec <--> CPU */ @@ -226,7 +223,11 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .ops = &broadwell_rt286_ops, .dpcm_playback = 1, .dpcm_capture = 1, +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) + SND_SOC_DAILINK_REG(dummy, codec, dummy), +#else SND_SOC_DAILINK_REG(ssp0_port, codec, platform), +#endif }, }; @@ -283,7 +284,7 @@ static int broadwell_audio_probe(struct platform_device *pdev) broadwell_rt286.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&broadwell_rt286, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 9177401c37a5..44016c16f25e 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -179,8 +179,8 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int clk_freq; /* Configure sysclk for codec */ @@ -226,7 +226,7 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct bxt_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -244,7 +244,7 @@ static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -721,7 +721,7 @@ static int broxton_audio_probe(struct platform_device *pdev) } /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 4b67f261377c..7a4decf34191 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -155,7 +155,7 @@ static const struct snd_soc_dapm_route geminilake_rt298_map[] = { static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -165,7 +165,7 @@ static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd) static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new(rtd->card, "Headset", @@ -186,7 +186,7 @@ static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct bxt_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -225,7 +225,7 @@ static int broxton_rt298_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, @@ -627,7 +627,7 @@ static int broxton_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 01739ad75b12..f5097da28828 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -89,7 +89,7 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) card->dapm.idle_bias_off = true; - ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), M98090_REG_SYSTEM_CLOCK, 25000000, SND_SOC_CLOCK_IN); if (ret < 0) { diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index 0c76dafdd572..ace232f8aed6 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -73,7 +73,7 @@ static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, @@ -123,7 +123,7 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; struct snd_soc_card *card = runtime->card; const struct snd_soc_dapm_route *custom_map; int num_routes; diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index 67f06c95eec5..3b3df7c9008c 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -70,7 +70,7 @@ static const struct acpi_gpio_mapping byt_cht_cx2072x_acpi_gpios[] = { static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_component *codec = rtd->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; int ret; if (devm_acpi_dev_add_driver_gpios(codec->dev, @@ -80,7 +80,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) card->dapm.idle_bias_off = true; /* set the default PLL rate, the clock is handled by the codec driver */ - ret = snd_soc_dai_set_sysclk(rtd->codec_dai, CX2072X_MCLK_EXTERNAL_PLL, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), CX2072X_MCLK_EXTERNAL_PLL, 19200000, SND_SOC_CLOCK_IN); if (ret) { dev_err(rtd->dev, "Could not set sysclk\n"); @@ -97,7 +97,7 @@ static int byt_cht_cx2072x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_component_set_jack(codec, &byt_cht_cx2072x_headset, NULL); - snd_soc_dai_set_bclk_ratio(rtd->codec_dai, 50); + snd_soc_dai_set_bclk_ratio(asoc_rtd_to_codec(rtd, 0), 50); return ret; } @@ -123,7 +123,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -132,7 +132,7 @@ static int byt_cht_cx2072x_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index eda7a500cad6..5e96e7d02733 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -78,7 +78,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -87,7 +87,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; @@ -106,7 +106,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK, @@ -127,7 +127,7 @@ static int aif1_hw_params(struct snd_pcm_substream *substream, static int aif1_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_pll(codec_dai, 0, @@ -231,7 +231,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) int ret_val = 0; int i; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; card = &bytcht_da7213_card; card->dev = &pdev->dev; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 0adc5a5e134a..ddcd070100ef 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -157,7 +157,7 @@ static struct snd_soc_jack_pin byt_cht_es8316_jack_pins[] = { static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_component *codec = runtime->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component; struct snd_soc_card *card = runtime->card; struct byt_cht_es8316_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; @@ -212,7 +212,7 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime) if (ret) dev_err(card->dev, "unable to enable MCLK\n"); - ret = snd_soc_dai_set_sysclk(runtime->codec_dai, 0, 19200000, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(runtime, 0), 0, 19200000, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(card->dev, "can't set codec clock %d\n", ret); @@ -262,7 +262,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -272,7 +272,7 @@ static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c index 479af808ef43..8c0dab1f4030 100644 --- a/sound/soc/intel/boards/bytcht_nocodec.c +++ b/sound/soc/intel/boards/bytcht_nocodec.c @@ -58,7 +58,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 24-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -68,7 +68,7 @@ static int codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 24); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6bd9ae813be2..33fb8ea4e5cb 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -381,7 +381,7 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); return byt_rt5640_prepare_and_enable_pll1(dai, params_rate(params)); } @@ -805,7 +805,7 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map; int num_routes; int ret; @@ -962,7 +962,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -971,7 +971,7 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 5074bb53f98e..214ef41e23e6 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -348,7 +348,7 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); snd_pcm_format_t format = params_format(params); int rate = params_rate(params); int bclk_ratio; @@ -540,7 +540,7 @@ static int byt_rt5651_add_codec_device_props(struct device *i2c_dev) static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *codec = runtime->codec_dai->component; + struct snd_soc_component *codec = asoc_rtd_to_codec(runtime, 0)->component; struct byt_rt5651_private *priv = snd_soc_card_get_drvdata(card); const struct snd_soc_dapm_route *custom_map; int num_routes; @@ -685,7 +685,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -696,7 +696,7 @@ static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, bits); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, bits); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70bb86f3342f..135701738a44 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -113,7 +113,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK, @@ -257,7 +257,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, int ret = 0; unsigned int fmt = 0; - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16); if (ret < 0) { dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret); return ret; @@ -266,7 +266,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), fmt); if (ret < 0) { dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret); return ret; @@ -553,7 +553,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 501bad3976fb..f456150f89c2 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -73,7 +73,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, NAU8824_CLK_FLL_FS, 0, @@ -96,7 +96,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); struct snd_soc_jack *jack = &ctx->jack; - struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); struct snd_soc_component *component = codec_dai->component; int ret, jack_type; @@ -259,7 +259,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* override plaform name, if required */ snd_soc_card_cht.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret_val = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cht, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index b5b016d493f1..e64eca56e426 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -208,7 +208,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ @@ -252,7 +252,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int jack_type; int ret; @@ -359,7 +359,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, * with explicit setting to I2S 2ch 16-bit. The word length is set with * dai_set_tdm_slot() since there is no other API exposed */ - ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -369,7 +369,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS @@ -379,7 +379,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } - ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2, 16); if (ret < 0) { dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); return ret; @@ -393,7 +393,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* * Default mode for SSP configuration is TDM 4 slot */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -403,7 +403,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); if (ret < 0) { dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); return ret; @@ -539,7 +539,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (!drv) return -ENOMEM; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { if (acpi_dev_found(snd_soc_cards[i].codec_id) && diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 9d657421730a..097023a3ec14 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -144,7 +144,7 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ @@ -176,7 +176,7 @@ static const struct acpi_gpio_mapping cht_rt5672_gpios[] = { static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); struct snd_soc_component *component = codec_dai->component; struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); @@ -255,7 +255,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* * Default mode for SSP configuration is TDM 4 slot */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -265,7 +265,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, } /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); if (ret < 0) { dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d0186a6c..8167b2977e1d 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -85,7 +85,7 @@ static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; int ret; @@ -125,7 +125,7 @@ static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_id, clk_freq, pll_out, ret; clk_id = RT5682_PLL1_S_MCLK; @@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* 100 Fs to drive 24 bit data */ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, @@ -275,7 +274,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) static int hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -447,12 +446,12 @@ static int snd_cml_rt1011_probe(struct platform_device *pdev) const char *platform_name; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; snd_soc_card_cml.dev = &pdev->dev; platform_name = mach->mach_params.platform; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 8e947bad143c..f13158e4a1fc 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -136,8 +136,8 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; int ret; @@ -188,7 +188,7 @@ static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* Set valid bitmask & configuration for I2S in 24 bit */ @@ -208,7 +208,7 @@ static struct snd_soc_ops geminilake_rt5682_ops = { static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct glk_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -225,7 +225,7 @@ static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; struct snd_soc_dapm_context *dapm; int ret; @@ -409,6 +409,7 @@ static struct snd_soc_dai_link geminilake_dais[] = { .init = NULL, .capture_only = 1, .nonatomic = 1, + .dynamic = 1, SND_SOC_DAILINK_REG(echoref, dummy, platform), }, [GLK_DPCM_AUDIO_REF_CP] = { @@ -604,7 +605,7 @@ static int geminilake_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; platform_name = mach->mach_params.platform; ret = snd_soc_fixup_dai_links_platform_name(card, platform_name); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 3dadf9bff796..3ed53d7db4e6 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -56,7 +56,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, @@ -193,7 +193,7 @@ static int haswell_audio_probe(struct platform_device *pdev) haswell_rt5640.dev = &pdev->dev; /* override plaform name, if required */ - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(&haswell_rt5640, mach->mach_params.platform); if (ret) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index bc7f9a9ce9af..32cd90b8d4c4 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -159,8 +159,8 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_jack *jack; int ret; @@ -203,7 +203,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -236,7 +236,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 7a13e9b35187..abd4e3839678 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai; int ret, j; - for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + for_each_rtd_codec_dais(runtime, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); @@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int j, ret; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dais(rtd, j, codec_dai) { const char *name = codec_dai->component->name; struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = @@ -331,7 +331,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; struct snd_soc_card *card = rtd->card; int ret; @@ -381,7 +381,7 @@ static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -414,7 +414,7 @@ static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index e23dea9ab79a..6460e3f0c974 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -157,7 +157,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); @@ -210,7 +210,7 @@ static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -244,7 +244,7 @@ static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index d8f2ff7139a9..658a9da3a40f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -242,7 +242,7 @@ static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -258,7 +258,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; /* @@ -305,7 +305,7 @@ static int kabylake_rt5663_max98927_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -431,7 +431,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ @@ -472,7 +472,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret = 0, j; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* * Use channel 4 and 5 for the first amp @@ -962,7 +962,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 96c814f36458..1b1f8d7a4ea3 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -206,7 +206,7 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = { static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; int ret; dapm = snd_soc_component_get_dapm(component); @@ -221,7 +221,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; /* @@ -255,7 +255,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct kbl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -372,7 +372,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ @@ -399,7 +399,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int ret = 0, j; - for_each_rtd_codec_dai(rtd, j, codec_dai) { + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); if (ret < 0) { @@ -772,7 +772,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&kabylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index d6150670ca05..e8545d13062f 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -49,6 +49,10 @@ static inline int skl_hda_hdmi_build_controls(struct snd_soc_card *card) struct snd_soc_component *component; struct skl_hda_hdmi_pcm *pcm; + /* HDMI disabled, do not create controls */ + if (list_empty(&ctx->hdmi_pcm_list)) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm, head); component = pcm->codec_dai->component; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 11eaee9ae41f..3be764299ab0 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -61,6 +61,9 @@ static const struct snd_soc_dapm_route skl_hda_map[] = { { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, }; +SND_SOC_DAILINK_DEF(dummy_codec, + DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai"))); + static int skl_hda_card_late_probe(struct snd_soc_card *card) { return skl_hda_hdmi_jack_init(card); @@ -114,13 +117,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) { struct snd_soc_card *card = &hda_soc_card; struct snd_soc_dai_link *dai_link; - u32 codec_count, codec_mask; + u32 codec_count, codec_mask, idisp_mask; int i, num_links, num_route; codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); + idisp_mask = codec_mask & IDISP_CODEC_MASK; + + if (!codec_count || codec_count > 2 || + (codec_count == 2 && !idisp_mask)) + return -EINVAL; - if (codec_count == 1 && codec_mask & IDISP_CODEC_MASK) { + if (codec_mask == idisp_mask) { + /* topology with iDisp as the only HDA codec */ num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; @@ -135,13 +144,19 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) skl_hda_be_dai_links[IDISP_DAI_COUNT + HDAC_DAI_COUNT + i]; } - } else if (codec_count == 2 && codec_mask & IDISP_CODEC_MASK) { + } else { + /* topology with external and iDisp HDA codecs */ num_links = ARRAY_SIZE(skl_hda_be_dai_links); num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - } else { - return -EINVAL; + if (!idisp_mask) { + for (i = 0; i < IDISP_DAI_COUNT; i++) { + skl_hda_be_dai_links[i].codecs = dummy_codec; + skl_hda_be_dai_links[i].num_codecs = + ARRAY_SIZE(dummy_codec); + } + } } card->num_links = num_links; @@ -167,7 +182,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (!mach) return -EINVAL; diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index e6de3b28d840..d7b8154c43a4 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -157,7 +157,7 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* * Headset buttons map to the google Reference headset. @@ -182,7 +182,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -200,7 +200,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -218,7 +218,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -236,7 +236,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -296,7 +296,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, @@ -660,7 +660,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index c99c8b23e509..4b317bcf6ea0 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -161,12 +161,12 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; /* Slot 1 for left */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[0], 0x01, 0x01, 2, 48); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0x01, 0x01, 2, 48); if (ret < 0) return ret; /* Slot 2 for right */ - ret = snd_soc_dai_set_tdm_slot(rtd->codec_dais[1], 0x02, 0x02, 2, 48); + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 1), 0x02, 0x02, 2, 48); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) { int ret; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* * 4 buttons here map to the google Reference headset @@ -201,7 +201,7 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -219,7 +219,7 @@ static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -238,7 +238,7 @@ static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -256,7 +256,7 @@ static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -348,7 +348,7 @@ static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, @@ -686,6 +686,7 @@ static struct snd_soc_card skylake_audio_card = { .codec_conf = ssm4567_codec_conf, .num_configs = ARRAY_SIZE(ssm4567_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; @@ -703,7 +704,7 @@ static int skylake_audio_probe(struct platform_device *pdev) skylake_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; if (mach) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index a9aec66a2351..903ae1b28ec9 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -112,7 +112,7 @@ static const struct snd_soc_dapm_route skylake_rt286_map[] = { static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_component *component = asoc_rtd_to_cpu(rtd, 0)->component; dapm = snd_soc_component_get_dapm(component); snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); @@ -122,7 +122,7 @@ static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd) static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; ret = snd_soc_card_jack_new(rtd->card, "Headset", @@ -143,7 +143,7 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct skl_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -229,7 +229,7 @@ static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 8f44f13d2848..b707dd3b5625 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -2,7 +2,7 @@ // Copyright(c) 2019 Intel Corporation. /* - * Intel SOF Machine driver for DA7219 + MAX98373 codec + * Intel SOF Machine driver for DA7219 + MAX98373/MAX98360A codec */ #include <linux/input.h> @@ -69,11 +69,20 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new m98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +/* For MAX98373 amp */ static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), @@ -83,21 +92,45 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "HPL" }, { "Headphone Jack", NULL, "HPR" }, + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, + { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, +}; + +/* For MAX98360A amp */ +static const struct snd_soc_dapm_widget max98360a_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_SPK("Spk", NULL), + + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_route max98360a_map[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, { "MIC", NULL, "Headset Mic" }, { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, + + {"Spk", NULL, "Speaker"}, }; static struct snd_soc_jack headset; static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; struct snd_soc_jack *jack; int ret; @@ -140,7 +173,7 @@ static int ssp1_hw_params(struct snd_pcm_substream *substream, int ret, j; for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, j); if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { /* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */ @@ -181,7 +214,7 @@ static struct snd_soc_codec_conf max98373_codec_conf[] = { static int hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -224,6 +257,9 @@ SND_SOC_DAILINK_DEF(ssp1_amps, /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI), /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI))); +SND_SOC_DAILINK_DEF(ssp1_m98360a, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi"))); + SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); SND_SOC_DAILINK_DEF(dmic_codec, @@ -320,6 +356,21 @@ static struct snd_soc_card card_da7219_m98373 = { .late_probe = card_late_probe, }; +static struct snd_soc_card card_da7219_m98360a = { + .name = "da7219max98360a", + .owner = THIS_MODULE, + .dai_link = dais, + .num_links = ARRAY_SIZE(dais), + .controls = m98360a_controls, + .num_controls = ARRAY_SIZE(m98360a_controls), + .dapm_widgets = max98360a_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98360a_widgets), + .dapm_routes = max98360a_map, + .num_dapm_routes = ARRAY_SIZE(max98360a_map), + .fully_routed = true, + .late_probe = card_late_probe, +}; + static int audio_probe(struct platform_device *pdev) { static struct snd_soc_card *card; @@ -327,15 +378,26 @@ static int audio_probe(struct platform_device *pdev) struct card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + /* By default dais[0] is configured for max98373 */ + if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + dais[0] = (struct snd_soc_dai_link) { + .name = "SSP1-Codec", + .id = 0, + .no_pcm = 1, + .dpcm_playback = 1, + .ignore_pmdown_time = 1, + SND_SOC_DAILINK_REG(ssp1_pin, ssp1_m98360a, platform) }; + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); card = (struct snd_soc_card *)pdev->id_entry->driver_data; card->dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); if (ret) @@ -351,13 +413,17 @@ static const struct platform_device_id board_ids[] = { .name = "sof_da7219_max98373", .driver_data = (kernel_ulong_t)&card_da7219_m98373, }, + { + .name = "sof_da7219_max98360a", + .driver_data = (kernel_ulong_t)&card_da7219_m98360a, + }, { } }; static struct platform_driver audio = { .probe = audio_probe, .driver = { - .name = "sof_da7219_max98373", + .name = "sof_da7219_max98_360a_373", .pm = &snd_soc_pm_ops, }, .id_table = board_ids, @@ -368,4 +434,5 @@ module_platform_driver(audio) MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver"); MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_da7219_max98360a"); MODULE_ALIAS("platform:sof_da7219_max98373"); diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c new file mode 100644 index 000000000000..463b39a7ccfd --- /dev/null +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -0,0 +1,80 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2020 Intel Corporation. All rights reserved. +#include <linux/string.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <uapi/sound/asound.h> +#include "sof_maxim_common.h" + +static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98373_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98373_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98373_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98373_components[] = { + { /* For Left */ + .name = MAX_98373_DEV0_NAME, + .dai_name = MAX_98373_CODEC_DAI, + }, + { /* For Right */ + .name = MAX_98373_DEV1_NAME, + .dai_name = MAX_98373_CODEC_DAI, + }, +}; + +static int max98373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + } + if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + } + } + return 0; +} + +struct snd_soc_ops max_98373_ops = { + .hw_params = max98373_hw_params, +}; + +int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +void sof_max98373_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98373_codec_conf; + card->num_configs = ARRAY_SIZE(max_98373_codec_conf); +} diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h new file mode 100644 index 000000000000..406bf0e81155 --- /dev/null +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2020 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with Maxim Codecs. + */ +#ifndef __SOF_MAXIM_COMMON_H +#define __SOF_MAXIM_COMMON_H + +#include <sound/soc.h> + +#define MAX_98373_CODEC_DAI "max98373-aif1" +#define MAX_98373_DEV0_NAME "i2c-MX98373:00" +#define MAX_98373_DEV1_NAME "i2c-MX98373:01" + +extern struct snd_soc_dai_link_component max_98373_components[2]; +extern struct snd_soc_ops max_98373_ops; + +int max98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); +void sof_max98373_codec_conf(struct snd_soc_card *card); +#endif /* __SOF_MAXIM_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c new file mode 100644 index 000000000000..fb7811899999 --- /dev/null +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -0,0 +1,448 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018-2020 Intel Corporation. + +/* + * Intel SOF Machine Driver for Intel platforms with TI PCM512x codec, + * e.g. Up or Up2 with Hifiberry DAC+ HAT + */ +#include <linux/clk.h> +#include <linux/dmi.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/types.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../codecs/pcm512x.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" + +#define NAME_SIZE 32 + +#define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) +#define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) + +#define IDISP_CODEC_MASK 0x4 + +/* Default: SSP5 */ +static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); + +static bool is_legacy_cpu; + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct list_head hdmi_pcm_list; + bool idisp_codec; +}; + +static int sof_pcm512x_quirk_cb(const struct dmi_system_id *id) +{ + sof_pcm512x_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_pcm512x_quirk_table[] = { + { + .callback = sof_pcm512x_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"), + }, + .driver_data = (void *)(SOF_PCM512X_SSP_CODEC(2)), + }, + {} +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_pcm512x_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_EN, 0x08, 0x08); + snd_soc_component_update_bits(codec, PCM512x_GPIO_OUTPUT_4, 0x0f, 0x02); + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x08); + + return 0; +} + +static int aif1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x08); + + return 0; +} + +static void aif1_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_update_bits(codec, PCM512x_GPIO_CONTROL_1, + 0x08, 0x00); +} + +static const struct snd_soc_ops sof_pcm512x_ops = { + .startup = aif1_startup, + .shutdown = aif1_shutdown, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct sof_hdmi_pcm *pcm; + + /* HDMI is not supported by SOF on Baytrail/CherryTrail */ + if (is_legacy_cpu) + return 0; + + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + if (!ctx->idisp_codec) + return 0; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); +} +#else +static int sof_card_late_probe(struct snd_soc_card *card) +{ + return 0; +} +#endif + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* Speaker */ + {"Ext Spk", NULL, "OUTR"}, + {"Ext Spk", NULL, "OUTL"}, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for pcm512x codec */ +static struct snd_soc_card sof_audio_card_pcm512x = { + .name = "pcm512x", + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +SND_SOC_DAILINK_DEF(pcm512x_component, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-104C5122:00", "pcm512x-hifi"))); +SND_SOC_DAILINK_DEF(dmic_component, + DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int dmic_be_num, + int hdmi_num, + bool idisp_codec) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links, + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + cpus = devm_kcalloc(dev, sof_audio_card_pcm512x.num_links, + sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = pcm512x_component; + links[id].num_codecs = ARRAY_SIZE(pcm512x_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_pcm512x_codec_init; + links[id].ops = &sof_pcm512x_ops; + links[id].nonatomic = true; + links[id].dpcm_playback = 1; + /* + * capture only supported with specific versions of the Hifiberry DAC+ + * links[id].dpcm_capture = 1; + */ + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + if (is_legacy_cpu) { + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "ssp%d-port", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + } else { + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + } + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kcalloc(dev, hdmi_num, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + /* + * topology cannot be loaded if codec is missing, so + * use the dummy codec if needed + */ + if (idisp_codec) { + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = + devm_kasprintf(dev, GFP_KERNEL, + "intel-hdmi-hifi%d", i); + } else { + idisp_components[i - 1].name = "snd-soc-dummy"; + idisp_components[i - 1].dai_name = "snd-soc-dummy-dai"; + } + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; + struct snd_soc_dai_link *dai_links; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + hdmi_num = 0; + if (soc_intel_is_byt() || soc_intel_is_cht()) { + is_legacy_cpu = true; + dmic_be_num = 0; + /* default quirk for legacy cpu */ + sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(2); + } else { + dmic_be_num = 2; +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) + if (mach->mach_params.common_hdmi_codec_drv && + (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) + ctx->idisp_codec = true; + + /* links are always present in topology */ + hdmi_num = 3; +#endif + } + + dmi_check_system(sof_pcm512x_quirk_table); + + dev_dbg(&pdev->dev, "sof_pcm512x_quirk = %lx\n", sof_pcm512x_quirk); + + ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, + dmic_be_num, hdmi_num, + ctx->idisp_codec); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_pcm512x.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_pcm512x.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_pcm512x, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_pcm512x, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_pcm512x); +} + +static int sof_pcm512x_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_component *component = NULL; + + for_each_card_components(card, component) { + if (!strcmp(component->name, pcm512x_component[0].name)) { + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .remove = sof_pcm512x_remove, + .driver = { + .name = "sof_pcm512x", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(sof_audio) + +MODULE_DESCRIPTION("ASoC Intel(R) SOF + PCM512x Machine driver"); +MODULE_AUTHOR("Pierre-Louis Bossart"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_pcm512x"); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d878873a8e0..8c29431b5847 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1,9 +1,9 @@ // SPDX-License-Identifier: GPL-2.0 -// Copyright(c) 2019 Intel Corporation. +// Copyright(c) 2019-2020 Intel Corporation. /* * Intel SOF Machine Driver with Realtek rt5682 Codec - * and speaker codec MAX98357A + * and speaker codec MAX98357A or RT1015. */ #include <linux/i2c.h> #include <linux/input.h> @@ -18,10 +18,12 @@ #include <sound/soc.h> #include <sound/rt5682.h> #include <sound/soc-acpi.h> +#include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" +#include "sof_maxim_common.h" #define NAME_SIZE 32 @@ -39,6 +41,8 @@ #define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) #define SOF_RT5682_NUM_HDMIDEV(quirk) \ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) +#define SOF_RT1015_SPEAKER_AMP_PRESENT BIT(13) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(14) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -120,7 +124,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct sof_hdmi_pcm *pcm; pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); @@ -139,7 +143,7 @@ static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; int ret; @@ -207,7 +211,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_id, clk_freq, pll_out, ret; if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { @@ -260,6 +264,42 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; +static int sof_rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai; + int i, ret; + + if (!snd_soc_card_get_codec_dai(card, "rt1015-aif")) + return 0; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1015_PLL_S_BCLK, + params_rate(params) * 50, + params_rate(params) * 256); + if (ret < 0) { + dev_err(card->dev, "failed to set pll\n"); + return ret; + } + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT1015_SCLK_S_PLL, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "failed to set sysclk\n"); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops sof_rt1015_ops = { + .hw_params = sof_rt1015_hw_params, +}; + static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -316,12 +356,17 @@ static const struct snd_kcontrol_new sof_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + }; static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_widget dmic_widgets[] = { @@ -342,11 +387,22 @@ static const struct snd_soc_dapm_route speaker_map[] = { { "Spk", NULL, "Speaker" }, }; +static const struct snd_soc_dapm_route speaker_map_lr[] = { + { "Left Spk", NULL, "Left SPO" }, + { "Right Spk", NULL, "Right SPO" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) +{ + return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); +} + static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -382,6 +438,17 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static struct snd_soc_codec_conf rt1015_amp_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-10EC1015:01"), + .name_prefix = "Right", + }, +}; + /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "rt5682", /* the sof- prefix is added by the core */ @@ -417,6 +484,17 @@ static struct snd_soc_dai_link_component max98357a_component[] = { } }; +static struct snd_soc_dai_link_component rt1015_components[] = { + { + .name = "i2c-10EC1015:00", + .dai_name = "rt1015-aif", + }, + { + .name = "i2c-10EC1015:01", + .dai_name = "rt1015-aif", + }, +}; + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, @@ -556,11 +634,24 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; links[id].id = id; - links[id].codecs = max98357a_component; - links[id].num_codecs = ARRAY_SIZE(max98357a_component); + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1015_components; + links[id].num_codecs = ARRAY_SIZE(rt1015_components); + links[id].init = speaker_codec_init_lr; + links[id].ops = &sof_rt1015_ops; + } else if (sof_rt5682_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max98373_spk_codec_init; + links[id].ops = &max_98373_ops; + } else { + links[id].codecs = max98357a_component; + links[id].num_codecs = ARRAY_SIZE(max98357a_component); + links[id].init = speaker_codec_init; + } links[id].platforms = platform_component; links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].init = speaker_codec_init, links[id].nonatomic = true; links[id].dpcm_playback = 1; links[id].no_pcm = 1; @@ -604,7 +695,7 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); - mach = (&pdev->dev)->platform_data; + mach = pdev->dev.platform_data; /* A speaker amp might not be present when the quirk claims one is. * Detect this via whether the machine driver match includes quirk_data. @@ -662,6 +753,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_rt5682.num_links++; + if (sof_rt5682_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + sof_max98373_codec_conf(&sof_audio_card_rt5682); + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, dmic_be_num, hdmi_num); if (!dai_links) @@ -669,6 +763,11 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_rt5682.dai_link = dai_links; + if (sof_rt5682_quirk & SOF_RT1015_SPEAKER_AMP_PRESENT) { + sof_audio_card_rt5682.codec_conf = rt1015_amp_conf; + sof_audio_card_rt5682.num_configs = ARRAY_SIZE(rt1015_amp_conf); + } + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; @@ -714,6 +813,24 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1015", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, + { + .name = "tgl_max98373_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; @@ -735,3 +852,5 @@ MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); MODULE_ALIAS("platform:tgl_max98357a_rt5682"); +MODULE_ALIAS("platform:jsl_rt5682_rt1015"); +MODULE_ALIAS("platform:tgl_max98373_rt5682"); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c new file mode 100644 index 000000000000..a64dc563b47e --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw.c @@ -0,0 +1,962 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw - ASOC Machine driver for Intel SoundWire platforms + */ + +#include <linux/device.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "sof_sdw_common.h" + +unsigned long sof_sdw_quirk = SOF_RT711_JD_SRC_JD1; + +#define INC_ID(BE, CPU, LINK) do { (BE)++; (CPU)++; (LINK)++; } while (0) + +static int sof_sdw_quirk_cb(const struct dmi_system_id *id) +{ + sof_sdw_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX), + }, + { + /* early version of SKU 09C6 */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, + "Tiger Lake Client Platform"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | + SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | + SOF_SSP_PORT(SOF_I2S_SSP2)), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)SOF_SDW_PCH_DMIC, + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "CometLake Client"), + }, + .driver_data = (void *)SOF_SDW_PCH_DMIC, + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC), + }, + + {} +}; + +static struct snd_soc_codec_conf codec_conf[] = { + { + .dlc = COMP_CODEC_CONF("sdw:0:25d:711:0"), + .name_prefix = "rt711", + }, + /* rt1308 w/ I2S connection */ + { + .dlc = COMP_CODEC_CONF("i2c-10EC1308:00"), + .name_prefix = "rt1308-1", + }, + /* rt1308 left on link 1 */ + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0"), + .name_prefix = "rt1308-1", + }, + /* two 1308s on link1 with different unique id */ + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:0"), + .name_prefix = "rt1308-1", + }, + { + .dlc = COMP_CODEC_CONF("sdw:1:25d:1308:0:2"), + .name_prefix = "rt1308-2", + }, + /* rt1308 right on link 2 */ + { + .dlc = COMP_CODEC_CONF("sdw:2:25d:1308:0"), + .name_prefix = "rt1308-2", + }, + { + .dlc = COMP_CODEC_CONF("sdw:3:25d:715:0"), + .name_prefix = "rt715", + }, + { + .dlc = COMP_CODEC_CONF("sdw:0:25d:5682:0"), + .name_prefix = "rt5682", + }, +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +/* these wrappers are only needed to avoid typecast compilation errors */ +static int sdw_startup(struct snd_pcm_substream *substream) +{ + return sdw_startup_stream(substream); +} + +static void sdw_shutdown(struct snd_pcm_substream *substream) +{ + sdw_shutdown_stream(substream); +} + +static const struct snd_soc_ops sdw_ops = { + .startup = sdw_startup, + .shutdown = sdw_shutdown, +}; + +static struct sof_sdw_codec_info codec_info_list[] = { + { + .id = 0x700, + .direction = {true, true}, + .dai_name = "rt700-aif1", + .init = sof_sdw_rt700_init, + }, + { + .id = 0x711, + .direction = {true, true}, + .dai_name = "rt711-aif1", + .init = sof_sdw_rt711_init, + }, + { + .id = 0x1308, + .acpi_id = "10EC1308", + .direction = {true, false}, + .dai_name = "rt1308-aif", + .ops = &sof_sdw_rt1308_i2s_ops, + .init = sof_sdw_rt1308_init, + }, + { + .id = 0x715, + .direction = {false, true}, + .dai_name = "rt715-aif2", + .init = sof_sdw_rt715_init, + }, + { + .id = 0x5682, + .direction = {true, true}, + .dai_name = "rt5682-sdw", + .init = sof_sdw_rt5682_init, + }, +}; + +static inline int find_codec_info_part(unsigned int part_id) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (part_id == codec_info_list[i].id) + break; + + if (i == ARRAY_SIZE(codec_info_list)) + return -EINVAL; + + return i; +} + +static inline int find_codec_info_acpi(const u8 *acpi_id) +{ + int i; + + if (!acpi_id[0]) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (!memcmp(codec_info_list[i].acpi_id, acpi_id, + ACPI_ID_LEN)) + break; + + if (i == ARRAY_SIZE(codec_info_list)) + return -EINVAL; + + return i; +} + +/* + * get BE dailink number and CPU DAI number based on sdw link adr. + * Since some sdw slaves may be aggregated, the CPU DAI number + * may be larger than the number of BE dailinks. + */ +static int get_sdw_dailink_info(const struct snd_soc_acpi_link_adr *links, + int *sdw_be_num, int *sdw_cpu_dai_num) +{ + const struct snd_soc_acpi_link_adr *link; + bool group_visited[SDW_MAX_GROUPS]; + bool no_aggregation; + int i; + + no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; + *sdw_cpu_dai_num = 0; + *sdw_be_num = 0; + + if (!links) + return -EINVAL; + + for (i = 0; i < SDW_MAX_GROUPS; i++) + group_visited[i] = false; + + for (link = links; link->num_adr; link++) { + const struct snd_soc_acpi_endpoint *endpoint; + int part_id, codec_index; + int stream; + u64 adr; + + adr = link->adr_d->adr; + part_id = SDW_PART_ID(adr); + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + endpoint = link->adr_d->endpoints; + + /* count DAI number for playback and capture */ + for_each_pcm_streams(stream) { + if (!codec_info_list[codec_index].direction[stream]) + continue; + + (*sdw_cpu_dai_num)++; + + /* count BE for each non-aggregated slave or group */ + if (!endpoint->aggregated || no_aggregation || + !group_visited[endpoint->group_id]) + (*sdw_be_num)++; + } + + if (endpoint->aggregated) + group_visited[endpoint->group_id] = true; + } + + return 0; +} + +static void init_dai_link(struct snd_soc_dai_link *dai_links, int be_id, + char *name, int playback, int capture, + struct snd_soc_dai_link_component *cpus, + int cpus_num, + struct snd_soc_dai_link_component *codecs, + int codecs_num, + int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops) +{ + dai_links->id = be_id; + dai_links->name = name; + dai_links->platforms = platform_component; + dai_links->num_platforms = ARRAY_SIZE(platform_component); + dai_links->nonatomic = true; + dai_links->no_pcm = 1; + dai_links->cpus = cpus; + dai_links->num_cpus = cpus_num; + dai_links->codecs = codecs; + dai_links->num_codecs = codecs_num; + dai_links->dpcm_playback = playback; + dai_links->dpcm_capture = capture; + dai_links->init = init; + dai_links->ops = ops; +} + +static bool is_unique_device(const struct snd_soc_acpi_link_adr *link, + unsigned int sdw_version, + unsigned int mfg_id, + unsigned int part_id, + unsigned int class_id, + int index_in_link + ) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int sdw1_version, mfg1_id, part1_id, class1_id; + u64 adr; + + /* skip itself */ + if (i == index_in_link) + continue; + + adr = link->adr_d[i].adr; + + sdw1_version = SDW_VERSION(adr); + mfg1_id = SDW_MFG_ID(adr); + part1_id = SDW_PART_ID(adr); + class1_id = SDW_CLASS_ID(adr); + + if (sdw_version == sdw1_version && + mfg_id == mfg1_id && + part_id == part1_id && + class_id == class1_id) + return false; + } + + return true; +} + +static int create_codec_dai_name(struct device *dev, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link_component *codec, + int offset) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int sdw_version, unique_id, mfg_id; + unsigned int link_id, part_id, class_id; + int codec_index, comp_index; + char *codec_str; + u64 adr; + + adr = link->adr_d[i].adr; + + sdw_version = SDW_VERSION(adr); + link_id = SDW_DISCO_LINK_ID(adr); + unique_id = SDW_UNIQUE_ID(adr); + mfg_id = SDW_MFG_ID(adr); + part_id = SDW_PART_ID(adr); + class_id = SDW_CLASS_ID(adr); + + comp_index = i + offset; + if (is_unique_device(link, sdw_version, mfg_id, part_id, + class_id, i)) { + codec_str = "sdw:%x:%x:%x:%x"; + codec[comp_index].name = + devm_kasprintf(dev, GFP_KERNEL, codec_str, + link_id, mfg_id, part_id, + class_id); + } else { + codec_str = "sdw:%x:%x:%x:%x:%x"; + codec[comp_index].name = + devm_kasprintf(dev, GFP_KERNEL, codec_str, + link_id, mfg_id, part_id, + class_id, unique_id); + } + + if (!codec[comp_index].name) + return -ENOMEM; + + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + codec[comp_index].dai_name = + codec_info_list[codec_index].dai_name; + } + + return 0; +} + +static int set_codec_init_func(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + bool playback) +{ + int i; + + for (i = 0; i < link->num_adr; i++) { + unsigned int part_id; + int codec_index; + + part_id = SDW_PART_ID(link->adr_d[i].adr); + codec_index = find_codec_info_part(part_id); + + if (codec_index < 0) + return codec_index; + + if (codec_info_list[codec_index].init) + codec_info_list[codec_index].init(link, dai_links, + &codec_info_list[codec_index], + playback); + } + + return 0; +} + +/* + * check endpoint status in slaves and gather link ID for all slaves in + * the same group to generate different CPU DAI. Now only support + * one sdw link with all slaves set with only single group id. + * + * one slave on one sdw link with aggregated = 0 + * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI + * + * two or more slaves on one sdw link with aggregated = 0 + * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs + * + * multiple links with multiple slaves with aggregated = 1 + * one sdw BE DAI <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs + */ +static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, + struct device *dev, int *cpu_dai_id, int *cpu_dai_num, + int *codec_num, int *group_id, + bool *group_generated) +{ + const struct snd_soc_acpi_adr_device *adr_d; + const struct snd_soc_acpi_link_adr *adr_next; + bool no_aggregation; + int index = 0; + + no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; + *codec_num = adr_link->num_adr; + adr_d = adr_link->adr_d; + + /* make sure the link mask has a single bit set */ + if (!is_power_of_2(adr_link->mask)) + return -EINVAL; + + cpu_dai_id[index++] = ffs(adr_link->mask) - 1; + if (!adr_d->endpoints->aggregated || no_aggregation) { + *cpu_dai_num = 1; + *group_id = 0; + return 0; + } + + *group_id = adr_d->endpoints->group_id; + + /* gather other link ID of slaves in the same group */ + for (adr_next = adr_link + 1; adr_next && adr_next->num_adr; + adr_next++) { + const struct snd_soc_acpi_endpoint *endpoint; + + endpoint = adr_next->adr_d->endpoints; + if (!endpoint->aggregated || + endpoint->group_id != *group_id) + continue; + + /* make sure the link mask has a single bit set */ + if (!is_power_of_2(adr_next->mask)) + return -EINVAL; + + if (index >= SDW_MAX_CPU_DAIS) { + dev_err(dev, " cpu_dai_id array overflows"); + return -EINVAL; + } + + cpu_dai_id[index++] = ffs(adr_next->mask) - 1; + *codec_num += adr_next->num_adr; + } + + /* + * indicate CPU DAIs for this group have been generated + * to avoid generating CPU DAIs for this group again. + */ + group_generated[*group_id] = true; + *cpu_dai_num = index; + + return 0; +} + +static int create_sdw_dailink(struct device *dev, int *be_index, + struct snd_soc_dai_link *dai_links, + int sdw_be_num, int sdw_cpu_dai_num, + struct snd_soc_dai_link_component *cpus, + const struct snd_soc_acpi_link_adr *link, + int *cpu_id, bool *group_generated) +{ + const struct snd_soc_acpi_link_adr *link_next; + struct snd_soc_dai_link_component *codecs; + int cpu_dai_id[SDW_MAX_CPU_DAIS]; + int cpu_dai_num, cpu_dai_index; + unsigned int part_id, group_id; + int codec_idx = 0; + int i = 0, j = 0; + int codec_index; + int codec_num; + int stream; + int ret; + int k; + + ret = get_slave_info(link, dev, cpu_dai_id, &cpu_dai_num, &codec_num, + &group_id, group_generated); + if (ret) + return ret; + + codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + /* generate codec name on different links in the same group */ + for (link_next = link; link_next && link_next->num_adr && + i < cpu_dai_num; link_next++) { + const struct snd_soc_acpi_endpoint *endpoints; + + endpoints = link_next->adr_d->endpoints; + if (group_id && (!endpoints->aggregated || + endpoints->group_id != group_id)) + continue; + + /* skip the link excluded by this processed group */ + if (cpu_dai_id[i] != ffs(link_next->mask) - 1) + continue; + + ret = create_codec_dai_name(dev, link_next, codecs, codec_idx); + if (ret < 0) + return ret; + + /* check next link to create codec dai in the processed group */ + i++; + codec_idx += link_next->num_adr; + } + + /* find codec info to create BE DAI */ + part_id = SDW_PART_ID(link->adr_d[0].adr); + codec_index = find_codec_info_part(part_id); + if (codec_index < 0) + return codec_index; + + cpu_dai_index = *cpu_id; + for_each_pcm_streams(stream) { + char *name, *cpu_name; + int playback, capture; + static const char * const sdw_stream_name[] = { + "SDW%d-Playback", + "SDW%d-Capture", + }; + + if (!codec_info_list[codec_index].direction[stream]) + continue; + + /* create stream name according to first link id */ + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream], cpu_dai_id[0]); + if (!name) + return -ENOMEM; + + /* + * generate CPU DAI name base on the sdw link ID and + * PIN ID with offset of 2 according to sdw dai driver. + */ + for (k = 0; k < cpu_dai_num; k++) { + cpu_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", cpu_dai_id[k], + j + SDW_INTEL_BIDIR_PDI_BASE); + if (!cpu_name) + return -ENOMEM; + + if (cpu_dai_index >= sdw_cpu_dai_num) { + dev_err(dev, "invalid cpu dai index %d", + cpu_dai_index); + return -EINVAL; + } + + cpus[cpu_dai_index++].dai_name = cpu_name; + } + + if (*be_index >= sdw_be_num) { + dev_err(dev, " invalid be dai index %d", *be_index); + return -EINVAL; + } + + if (*cpu_id >= sdw_cpu_dai_num) { + dev_err(dev, " invalid cpu dai index %d", *cpu_id); + return -EINVAL; + } + + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); + capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + init_dai_link(dai_links + *be_index, *be_index, name, + playback, capture, + cpus + *cpu_id, cpu_dai_num, + codecs, codec_num, + NULL, &sdw_ops); + + ret = set_codec_init_func(link, dai_links + (*be_index)++, + playback); + if (ret < 0) { + dev_err(dev, "failed to init codec %d", codec_index); + return ret; + } + + *cpu_id += cpu_dai_num; + j++; + } + + return 0; +} + +/* + * DAI link ID of SSP & DMIC & HDMI are based on last + * link ID used by sdw link. Since be_id may be changed + * in init func of sdw codec, it is not equal to be_id + */ +static inline int get_next_be_id(struct snd_soc_dai_link *links, + int be_id) +{ + return links[be_id - 1].id + 1; +} + +static int sof_card_dai_links_create(struct device *dev, + struct snd_soc_acpi_mach *mach, + struct snd_soc_card *card) +{ + int ssp_num, sdw_be_num = 0, hdmi_num = 0, dmic_num; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + struct snd_soc_dai_link_component *idisp_components; +#endif + struct snd_soc_dai_link_component *ssp_components; + struct snd_soc_acpi_mach_params *mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + struct snd_soc_dai_link_component *cpus; + bool group_generated[SDW_MAX_GROUPS]; + int ssp_codec_index, ssp_mask; + struct snd_soc_dai_link *links; + int num_links, link_id = 0; + char *name, *cpu_name; + int total_cpu_dai_num; + int sdw_cpu_dai_num; + int i, j, be_id = 0; + int cpu_id = 0; + int comp_num; + int ret; + + /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + codec_info_list[i].amp_num = 0; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + hdmi_num = sof_sdw_quirk & SOF_SDW_TGL_HDMI ? + SOF_TGL_HDMI_COUNT : SOF_PRE_TGL_HDMI_COUNT; +#endif + + ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); + /* + * on generic tgl platform, I2S or sdw mode is supported + * based on board rework. A ACPI device is registered in + * system only when I2S mode is supported, not sdw mode. + * Here check ACPI ID to confirm I2S is supported. + */ + ssp_codec_index = find_codec_info_acpi(mach->id); + ssp_num = ssp_codec_index >= 0 ? hweight_long(ssp_mask) : 0; + comp_num = hdmi_num + ssp_num; + + mach_params = &mach->mach_params; + ret = get_sdw_dailink_info(mach_params->links, + &sdw_be_num, &sdw_cpu_dai_num); + if (ret < 0) { + dev_err(dev, "failed to get sdw link info %d", ret); + return ret; + } + + /* enable dmic01 & dmic16k */ + dmic_num = (sof_sdw_quirk & SOF_SDW_PCH_DMIC) ? 2 : 0; + comp_num += dmic_num; + + dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d", sdw_be_num, ssp_num, + dmic_num, hdmi_num); + + /* allocate BE dailinks */ + num_links = comp_num + sdw_be_num; + links = devm_kcalloc(dev, num_links, sizeof(*links), GFP_KERNEL); + + /* allocated CPU DAIs */ + total_cpu_dai_num = comp_num + sdw_cpu_dai_num; + cpus = devm_kcalloc(dev, total_cpu_dai_num, sizeof(*cpus), + GFP_KERNEL); + + if (!links || !cpus) + return -ENOMEM; + + /* SDW */ + if (!sdw_be_num) + goto SSP; + + adr_link = mach_params->links; + if (!adr_link) + return -EINVAL; + + /* + * SoundWire Slaves aggregated in the same group may be + * located on different hardware links. Clear array to indicate + * CPU DAIs for this group have not been generated. + */ + for (i = 0; i < SDW_MAX_GROUPS; i++) + group_generated[i] = false; + + /* generate DAI links by each sdw link */ + for (; adr_link->num_adr; adr_link++) { + const struct snd_soc_acpi_endpoint *endpoint; + + endpoint = adr_link->adr_d->endpoints; + if (endpoint->aggregated && !endpoint->group_id) { + dev_err(dev, "invalid group id on link %x", + adr_link->mask); + continue; + } + + /* this group has been generated */ + if (endpoint->aggregated && + group_generated[endpoint->group_id]) + continue; + + ret = create_sdw_dailink(dev, &be_id, links, sdw_be_num, + sdw_cpu_dai_num, cpus, adr_link, + &cpu_id, group_generated); + if (ret < 0) { + dev_err(dev, "failed to create dai link %d", be_id); + return -ENOMEM; + } + } + + /* non-sdw DAI follows sdw DAI */ + link_id = be_id; + + /* get BE ID for non-sdw DAI */ + be_id = get_next_be_id(links, be_id); + +SSP: + /* SSP */ + if (!ssp_num) + goto DMIC; + + for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) { + struct sof_sdw_codec_info *info; + int playback, capture; + char *codec_name; + + if (!(ssp_mask & 0x1)) + continue; + + name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", i); + if (!name) + return -ENOMEM; + + cpu_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); + if (!cpu_name) + return -ENOMEM; + + ssp_components = devm_kzalloc(dev, sizeof(*ssp_components), + GFP_KERNEL); + if (!ssp_components) + return -ENOMEM; + + info = &codec_info_list[ssp_codec_index]; + codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", + info->acpi_id, j++); + if (!codec_name) + return -ENOMEM; + + ssp_components->name = codec_name; + ssp_components->dai_name = info->dai_name; + cpus[cpu_id].dai_name = cpu_name; + + playback = info->direction[SNDRV_PCM_STREAM_PLAYBACK]; + capture = info->direction[SNDRV_PCM_STREAM_CAPTURE]; + init_dai_link(links + link_id, be_id, name, + playback, capture, + cpus + cpu_id, 1, + ssp_components, 1, + NULL, info->ops); + + ret = info->init(NULL, links + link_id, info, 0); + if (ret < 0) + return ret; + + INC_ID(be_id, cpu_id, link_id); + } + +DMIC: + /* dmic */ + if (dmic_num > 0) { + cpus[cpu_id].dai_name = "DMIC01 Pin"; + init_dai_link(links + link_id, be_id, "dmic01", + 0, 1, // DMIC only supports capture + cpus + cpu_id, 1, + dmic_component, 1, + sof_sdw_dmic_init, NULL); + INC_ID(be_id, cpu_id, link_id); + + cpus[cpu_id].dai_name = "DMIC16k Pin"; + init_dai_link(links + link_id, be_id, "dmic16k", + 0, 1, // DMIC only supports capture + cpus + cpu_id, 1, + dmic_component, 1, + /* don't call sof_sdw_dmic_init() twice */ + NULL, NULL); + INC_ID(be_id, cpu_id, link_id); + } + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kcalloc(dev, hdmi_num, + sizeof(*idisp_components), + GFP_KERNEL); + if (!idisp_components) + return -ENOMEM; + } + + for (i = 0; i < hdmi_num; i++) { + name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i + 1); + if (!name) + return -ENOMEM; + + idisp_components[i].name = "ehdaudio0D2"; + idisp_components[i].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i + 1); + if (!idisp_components[i].dai_name) + return -ENOMEM; + + cpu_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i + 1); + if (!cpu_name) + return -ENOMEM; + + cpus[cpu_id].dai_name = cpu_name; + init_dai_link(links + link_id, be_id, name, + 1, 0, // HDMI only supports playback + cpus + cpu_id, 1, + idisp_components + i, 1, + sof_sdw_hdmi_init, NULL); + INC_ID(be_id, cpu_id, link_id); + } +#endif + + card->dai_link = links; + card->num_links = num_links; + + return 0; +} + +/* SoC card */ +static const char sdw_card_long_name[] = "Intel Soundwire SOF"; + +static struct snd_soc_card card_sof_sdw = { + .name = "soundwire", + .late_probe = sof_sdw_hdmi_card_late_probe, + .codec_conf = codec_conf, + .num_configs = ARRAY_SIZE(codec_conf), +}; + +static int mc_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &card_sof_sdw; + struct snd_soc_acpi_mach *mach; + struct mc_private *ctx; + int ret; + + dev_dbg(&pdev->dev, "Entry %s\n", __func__); + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + dmi_check_system(sof_sdw_quirk_table); + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); +#endif + + card->dev = &pdev->dev; + + mach = pdev->dev.platform_data; + ret = sof_card_dai_links_create(&pdev->dev, mach, + card); + if (ret < 0) + return ret; + + ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; + + snd_soc_card_set_drvdata(card, ctx); + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "cfg-spk:%d", + (sof_sdw_quirk & SOF_SDW_FOUR_SPK) ? 4 : 2); + if (!card->components) + return -ENOMEM; + + card->long_name = sdw_card_long_name; + + /* Register the card */ + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + platform_set_drvdata(pdev, card); + + return ret; +} + +static struct platform_driver sof_sdw_driver = { + .driver = { + .name = "sof_sdw", + .pm = &snd_soc_pm_ops, + }, + .probe = mc_probe, +}; + +module_platform_driver(sof_sdw_driver); + +MODULE_DESCRIPTION("ASoC SoundWire Generic Machine driver"); +MODULE_AUTHOR("Bard Liao <yung-chuan.liao@linux.intel.com>"); +MODULE_AUTHOR("Rander Wang <rander.wang@linux.intel.com>"); +MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sof_sdw"); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h new file mode 100644 index 000000000000..dd593ff3575b --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -0,0 +1,114 @@ +/* SPDX-License-Identifier: GPL-2.0 + * Copyright (c) 2020 Intel Corporation + */ + +/* + * sof_sdw_common.h - prototypes for common helpers + */ + +#ifndef SND_SOC_SOF_SDW_COMMON_H +#define SND_SOC_SOF_SDW_COMMON_H + +#include <linux/bits.h> +#include <linux/types.h> + +#define MAX_NO_PROPS 2 +#define MAX_HDMI_NUM 4 +#define SDW_DMIC_DAI_ID 4 +#define SDW_MAX_CPU_DAIS 16 +#define SDW_INTEL_BIDIR_PDI_BASE 2 + +/* 8 combinations with 4 links + unused group 0 */ +#define SDW_MAX_GROUPS 9 + +enum { + SOF_RT711_JD_SRC_JD1 = 1, + SOF_RT711_JD_SRC_JD2 = 2, +}; + +enum { + SOF_PRE_TGL_HDMI_COUNT = 3, + SOF_TGL_HDMI_COUNT = 4, +}; + +enum { + SOF_I2S_SSP0 = BIT(0), + SOF_I2S_SSP1 = BIT(1), + SOF_I2S_SSP2 = BIT(2), + SOF_I2S_SSP3 = BIT(3), + SOF_I2S_SSP4 = BIT(4), + SOF_I2S_SSP5 = BIT(5), +}; + +#define SOF_RT711_JDSRC(quirk) ((quirk) & GENMASK(1, 0)) +#define SOF_SDW_FOUR_SPK BIT(2) +#define SOF_SDW_TGL_HDMI BIT(3) +#define SOF_SDW_PCH_DMIC BIT(4) +#define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 5) +#define SOF_SSP_GET_PORT(quirk) (((quirk) >> 5) & GENMASK(5, 0)) +#define SOF_RT715_DAI_ID_FIX BIT(11) +#define SOF_SDW_NO_AGGREGATION BIT(12) + +struct sof_sdw_codec_info { + const int id; + int amp_num; + const u8 acpi_id[ACPI_ID_LEN]; + const bool direction[2]; // playback & capture support + const char *dai_name; + const struct snd_soc_ops *ops; + + int (*init)(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); +}; + +struct mc_private { + struct list_head hdmi_pcm_list; + bool common_hdmi_codec_drv; + struct snd_soc_jack sdw_headset; +}; + +extern unsigned long sof_sdw_quirk; + +/* generic HDMI support */ +int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); + +int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); + +/* DMIC support */ +int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); + +/* RT711 support */ +int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT700 support */ +int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT1308 support */ +extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops; + +int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT715 support */ +int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +/* RT5682 support */ +int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + +#endif diff --git a/sound/soc/intel/boards/sof_sdw_dmic.c b/sound/soc/intel/boards/sof_sdw_dmic.c new file mode 100644 index 000000000000..e92176bf0ad4 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_dmic.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_dmic - Helpers to handle dmic from generic machine driver + */ + +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + diff --git a/sound/soc/intel/boards/sof_sdw_hdmi.c b/sound/soc/intel/boards/sof_sdw_hdmi.c new file mode 100644 index 000000000000..c7b5612a39e6 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_hdmi.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_hdmi - Helpers to handle HDMI from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <linux/kernel.h> +#include <linux/list.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/jack.h> +#include "sof_sdw_common.h" +#include "../../codecs/hdac_hdmi.h" +#include "hda_dsp_common.h" + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) +static struct snd_soc_jack hdmi[MAX_HDMI_NUM]; + +struct hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +#define NAME_SIZE 32 +int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct hdmi_pcm *pcm; + struct snd_soc_component *component = NULL; + int err, i = 0; + char jack_name[NAME_SIZE]; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, + head); + component = pcm->codec_dai->component; + + if (ctx->common_hdmi_codec_drv) + return hda_dsp_hdmi_build_controls(card, component); + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &hdmi[i], + NULL, 0); + + if (err) + return err; + + err = snd_jack_add_new_kctl(hdmi[i].jack, + jack_name, SND_JACK_AVOUT); + if (err) + dev_warn(component->dev, "failed creating Jack kctl\n"); + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} +#else +int hdmi_card_late_probe(struct snd_soc_card *card) +{ + return 0; +} +#endif diff --git a/sound/soc/intel/boards/sof_sdw_rt1308.c b/sound/soc/intel/boards/sof_sdw_rt1308.c new file mode 100644 index 000000000000..321768e54d08 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt1308.c @@ -0,0 +1,151 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt1308 - Helpers to handle RT1308 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "sof_sdw_common.h" +#include "../../codecs/rt1308.h" + +static const struct snd_soc_dapm_widget rt1308_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* + * dapm routes for rt1308 will be registered dynamically according + * to the number of rt1308 used. The first two entries will be registered + * for one codec case, and the last two entries are also registered + * if two 1308s are used. + */ +static const struct snd_soc_dapm_route rt1308_map[] = { + { "Speaker", NULL, "rt1308-1 SPOL" }, + { "Speaker", NULL, "rt1308-1 SPOR" }, + { "Speaker", NULL, "rt1308-2 SPOL" }, + { "Speaker", NULL, "rt1308-2 SPOR" }, +}; + +static const struct snd_kcontrol_new rt1308_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int first_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s spk:rt1308", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt1308_controls, + ARRAY_SIZE(rt1308_controls)); + if (ret) { + dev_err(card->dev, "rt1308 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt1308_widgets, + ARRAY_SIZE(rt1308_widgets)); + if (ret) { + dev_err(card->dev, "rt1308 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map, 2); + if (ret) + dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret); + + return ret; +} + +static int second_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1308_map + 2, 2); + if (ret) + dev_err(rtd->dev, "failed to add second SPK map: %d\n", ret); + + return ret; +} + +static int all_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + ret = first_spk_init(rtd); + if (ret) + return ret; + + return second_spk_init(rtd); +} + +static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int clk_id, clk_freq, pll_out; + int err; + + clk_id = RT1308_PLL_S_MCLK; + clk_freq = 38400000; + + pll_out = params_rate(params) * 512; + + /* Set rt1308 pll */ + err = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (err < 0) { + dev_err(card->dev, "Failed to set RT1308 PLL: %d\n", err); + return err; + } + + /* Set rt1308 sysclk */ + err = snd_soc_dai_set_sysclk(codec_dai, RT1308_FS_SYS_S_PLL, pll_out, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "Failed to set RT1308 SYSCLK: %d\n", err); + return err; + } + + return 0; +} + +/* machine stream operations */ +struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { + .hw_params = rt1308_i2s_hw_params, +}; + +int sof_sdw_rt1308_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + info->amp_num++; + if (info->amp_num == 1) + dai_links->init = first_spk_init; + + if (info->amp_num == 2) { + /* + * if two 1308s are in one dai link, the init function + * in this dai link will be first set for the first speaker, + * and it should be reset to initialize all speakers when + * the second speaker is found. + */ + if (dai_links->init) + dai_links->init = all_spk_init; + else + dai_links->init = second_spk_init; + } + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c new file mode 100644 index 000000000000..5aa6211a1ed9 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -0,0 +1,126 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <linux/input.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/jack.h> +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt5682_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route rt5682_map[] = { + /*Headphones*/ + { "Headphone", NULL, "rt5682 HPOL" }, + { "Headphone", NULL, "rt5682 HPOR" }, + { "rt5682 IN1P", NULL, "Headset Mic" }, +}; + +static const struct snd_kcontrol_new rt5682_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_jack_pin rt5682_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt5682", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt5682_controls, + ARRAY_SIZE(rt5682_controls)); + if (ret) { + dev_err(card->dev, "rt5682 control addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets, + ARRAY_SIZE(rt5682_widgets)); + if (ret) { + dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map, + ARRAY_SIZE(rt5682_map)); + + if (ret) { + dev_err(card->dev, "rt5682 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt5682_jack_pins, + ARRAY_SIZE(rt5682_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + dai_links->init = rt5682_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c new file mode 100644 index 000000000000..2ee4e6910d7f --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -0,0 +1,125 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt700 - Helpers to handle RT700 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <linux/input.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/jack.h> +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt700_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route rt700_map[] = { + /* Headphones */ + { "Headphones", NULL, "HP" }, + { "Speaker", NULL, "SPK" }, + { "MIC2", NULL, "AMIC" }, +}; + +static const struct snd_kcontrol_new rt700_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("AMIC"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static struct snd_soc_jack_pin rt700_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt700", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt700_controls, + ARRAY_SIZE(rt700_controls)); + if (ret) { + dev_err(card->dev, "rt700 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets, + ARRAY_SIZE(rt700_widgets)); + if (ret) { + dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map, + ARRAY_SIZE(rt700_map)); + + if (ret) { + dev_err(card->dev, "rt700 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt700_jack_pins, + ARRAY_SIZE(rt700_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt700_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + dai_links->init = rt700_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c new file mode 100644 index 000000000000..2a4917e3d561 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -0,0 +1,156 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt711 - Helpers to handle RT711 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <linux/input.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/jack.h> +#include "sof_sdw_common.h" + +/* + * Note this MUST be called before snd_soc_register_card(), so that the props + * are in place before the codec component driver's probe function parses them. + */ +static int rt711_add_codec_device_props(const char *sdw_dev_name) +{ + struct property_entry props[MAX_NO_PROPS] = {}; + struct device *sdw_dev; + int ret; + + sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, sdw_dev_name); + if (!sdw_dev) + return -EPROBE_DEFER; + + if (SOF_RT711_JDSRC(sof_sdw_quirk)) { + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", + SOF_RT711_JDSRC(sof_sdw_quirk)); + } + + ret = device_add_properties(sdw_dev, props); + put_device(sdw_dev); + + return ret; +} + +static const struct snd_soc_dapm_widget rt711_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route rt711_map[] = { + /* Headphones */ + { "Headphone", NULL, "rt711 HP" }, + { "rt711 MIC2", NULL, "Headset Mic" }, +}; + +static const struct snd_kcontrol_new rt711_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_jack_pin rt711_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_jack *jack; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s hs:rt711", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt711_controls, + ARRAY_SIZE(rt711_controls)); + if (ret) { + dev_err(card->dev, "rt711 controls addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets, + ARRAY_SIZE(rt711_widgets)); + if (ret) { + dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map, + ARRAY_SIZE(rt711_map)); + + if (ret) { + dev_err(card->dev, "rt711 map addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sdw_headset, + rt711_jack_pins, + ARRAY_SIZE(rt711_jack_pins)); + if (ret) { + dev_err(rtd->card->dev, "Headset Jack creation failed: %d\n", + ret); + return ret; + } + + jack = &ctx->sdw_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) + dev_err(rtd->card->dev, "Headset Jack call-back failed: %d\n", + ret); + + return ret; +} + +int sof_sdw_rt711_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + int ret; + + /* + * headset should be initialized once. + * Do it with dai link for playback. + */ + if (!playback) + return 0; + + ret = rt711_add_codec_device_props("sdw:0:25d:711:0"); + if (ret < 0) + return ret; + + dai_links->init = rt711_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c new file mode 100644 index 000000000000..321e1cbc03ed --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt715.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2020 Intel Corporation + +/* + * sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver + */ + +#include <linux/device.h> +#include <linux/errno.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "sof_sdw_common.h" + +static int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:rt715", + card->components); + if (!card->components) + return -ENOMEM; + + return 0; +} + +int sof_sdw_rt715_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * DAI ID is fixed at SDW_DMIC_DAI_ID for 715 to + * keep sdw DMIC and HDMI setting static in UCM + */ + if (sof_sdw_quirk & SOF_RT715_DAI_ID_FIX) + dai_links->id = SDW_DMIC_DAI_ID; + + dai_links->init = rt715_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c index 4a5adae1d785..f5092bc48364 100644 --- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c @@ -65,7 +65,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[] = { }, { .id = "104C5122", - .drv_name = "bxt-pcm512x", + .drv_name = "sof_pcm512x", .sof_fw_filename = "sof-apl.ri", .sof_tplg_filename = "sof-apl-pcm512x.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index d0fb43c2b9f6..2752dc955733 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -174,6 +174,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cherrytrail_machines[] = { .sof_fw_filename = "sof-cht.ri", .sof_tplg_filename = "sof-cht-cx2072x.tplg", }, + { + .id = "104C5122", + .drv_name = "sof_pcm512x", + .sof_fw_filename = "sof-cht.ri", + .sof_tplg_filename = "sof-cht-src-50khz-pcm512x.tplg", + }, + #if IS_ENABLED(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) /* * This is always last in the table so that it is selected only when diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index f55634c4c2e8..bcedec6c6117 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -59,42 +59,112 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, }; -static const u64 rt1308_1_adr[] = { - 0x000110025D130800 +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, }; -static const u64 rt1308_2_adr[] = { - 0x000210025D130800 +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, }; -static const u64 rt715_3_adr[] = { - 0x000310025D071500 +static const struct snd_soc_acpi_adr_device rt700_1_adr[] = { + { + .adr = 0x000110025D070000, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_link_adr cml_rvp[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt700_1_adr), + .adr_d = rt700_1_adr, + }, + {} +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt715_3_adr[] = { + { + .adr = 0x000310025D071500, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr cml_3_in_1_default[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), - .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .num_adr = ARRAY_SIZE(rt1308_1_group1_adr), + .adr_d = rt1308_1_group1_adr, }, { .mask = BIT(2), - .num_adr = ARRAY_SIZE(rt1308_2_adr), - .adr = rt1308_2_adr, + .num_adr = ARRAY_SIZE(rt1308_2_group1_adr), + .adr_d = rt1308_2_group1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -103,17 +173,17 @@ static const struct snd_soc_acpi_link_adr cml_3_in_1_mono_amp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -122,7 +192,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { { .link_mask = 0xF, /* 4 active links required */ .links = cml_3_in_1_default, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg", }, @@ -134,13 +204,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[] = { */ .link_mask = 0xF, .links = cml_3_in_1_mono_amp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg", }, { .link_mask = 0x2, /* RT700 connected on Link1 */ - .drv_name = "sdw_rt700", + .links = cml_rvp, + .drv_name = "sof_sdw", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt700.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index 752733013d54..ef8500349f2f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -33,55 +33,112 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); -static const u64 rt700_0_adr[] = { - 0x000010025D070000 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device rt700_0_adr[] = { + { + .adr = 0x000010025D070000, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr icl_rvp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt700_0_adr), - .adr = rt700_0_adr, + .adr_d = rt700_0_adr, }, {} }; -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; -static const u64 rt1308_1_adr[] = { - 0x000110025D130800 +static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; -static const u64 rt1308_2_adr[] = { - 0x000210025D130800 +static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = { + { + .adr = 0x000110025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + } }; -static const u64 rt715_3_adr[] = { - 0x000310025D071500 +static const struct snd_soc_acpi_adr_device rt1308_2_group1_adr[] = { + { + .adr = 0x000210025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt715_3_adr[] = { + { + .adr = 0x000310025D071500, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr icl_3_in_1_default[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), - .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .num_adr = ARRAY_SIZE(rt1308_1_group1_adr), + .adr_d = rt1308_1_group1_adr, }, { .mask = BIT(2), - .num_adr = ARRAY_SIZE(rt1308_2_adr), - .adr = rt1308_2_adr, + .num_adr = ARRAY_SIZE(rt1308_2_group1_adr), + .adr_d = rt1308_2_group1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -90,17 +147,17 @@ static const struct snd_soc_acpi_link_adr icl_3_in_1_mono_amp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, { .mask = BIT(3), .num_adr = ARRAY_SIZE(rt715_3_adr), - .adr = rt715_3_adr, + .adr_d = rt715_3_adr, }, {} }; @@ -109,21 +166,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[] = { { .link_mask = 0xF, /* 4 active links required */ .links = icl_3_in_1_default, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715.tplg", }, { .link_mask = 0xB, /* 3 active links required */ .links = icl_3_in_1_mono_amp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt711-rt1308-rt715-mono.tplg", }, { .link_mask = 0x1, /* rt700 connected on link0 */ .links = icl_rvp, - .drv_name = "sdw_rt700", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-icl.ri", .sof_tplg_filename = "sof-icl-rt700.tplg", }, diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index ed2b125f6a11..4388a32718d8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -2,20 +2,50 @@ /* * soc-apci-intel-jsl-match.c - tables and support for JSL ACPI enumeration. * - * Copyright (c) 2019, Intel Corporation. + * Copyright (c) 2019-2020, Intel Corporation. * */ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +static struct snd_soc_acpi_codecs jsl_7219_98373_codecs = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + +static struct snd_soc_acpi_codecs rt1015_spk = { + .num_codecs = 1, + .codecs = {"10EC1015"} +}; + +/* + * When adding new entry to the snd_soc_acpi_intel_jsl_machines array, + * use .quirk_data member to distinguish different machine driver, + * and keep ACPI .id field unchanged for the common codec. + */ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { { .id = "DLGS7219", .drv_name = "sof_da7219_max98373", - .machine_quirk = snd_soc_acpi_codec_list, .sof_fw_filename = "sof-jsl.ri", .sof_tplg_filename = "sof-jsl-da7219.tplg", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &jsl_7219_98373_codecs, + }, + { + .id = "DLGS7219", + .drv_name = "sof_da7219_max98360a", + .sof_fw_filename = "sof-jsl.ri", + .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", + }, + { + .id = "10EC5682", + .drv_name = "jsl_rt5682_rt1015", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1015_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 5984dd151f3e..449d9d2286ae 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -14,20 +14,61 @@ static struct snd_soc_acpi_codecs tgl_codecs = { .codecs = {"MX98357A"} }; -static const u64 rt711_0_adr[] = { - 0x000010025D071100 +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, }; -static const u64 rt1308_1_adr[] = { - 0x000120025D130800, - 0x000122025D130800 +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000010025D071100, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = { + { + .adr = 0x000120025D130800, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + }, + { + .adr = 0x000122025D130800, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + } +}; + +static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = { + { + .adr = 0x000021025D568200, + .num_endpoints = 1, + .endpoints = &single_endpoint, + } }; static const struct snd_soc_acpi_link_adr tgl_i2s_rt1308[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, {} }; @@ -36,24 +77,38 @@ static const struct snd_soc_acpi_link_adr tgl_rvp[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt711_0_adr), - .adr = rt711_0_adr, + .adr_d = rt711_0_adr, }, { .mask = BIT(1), .num_adr = ARRAY_SIZE(rt1308_1_adr), - .adr = rt1308_1_adr, + .adr_d = rt1308_1_adr, }, {} }; +static const struct snd_soc_acpi_link_adr tgl_chromebook_base[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt5682_0_adr), + .adr_d = rt5682_0_adr, + }, + {} +}; + +static struct snd_soc_acpi_codecs tgl_max98373_amp = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC1308", - .drv_name = "rt711_rt1308", + .drv_name = "sof_sdw", .link_mask = 0x1, /* RT711 on SoundWire link0 */ .links = tgl_i2s_rt1308, .sof_fw_filename = "sof-tgl.ri", - .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", + .sof_tplg_filename = "sof-tgl-rt711-i2s-rt1308.tplg", }, { .id = "10EC5682", @@ -63,6 +118,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "tgl_max98373_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &tgl_max98373_amp, + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); @@ -72,10 +135,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { { .link_mask = 0x3, /* rt711 on link 0 and 2 rt1308s on link 1 */ .links = tgl_rvp, - .drv_name = "sdw_rt711_rt1308_rt715", + .drv_name = "sof_sdw", .sof_fw_filename = "sof-tgl.ri", .sof_tplg_filename = "sof-tgl-rt711-rt1308.tplg", }, + { + .link_mask = 0x1, /* this will only enable rt5682 for now */ + .links = tgl_chromebook_base, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_sdw_machines); diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 033d7c05d7fb..c183f8e94ee4 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -476,7 +476,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, u8 channels; int ret, dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ @@ -494,7 +494,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, } pcm_data->allocated = false; - pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id, hsw_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "error: failed to create stream\n"); @@ -509,7 +509,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, path_id = SST_HSW_STREAM_PATH_SSP0_IN; /* DSP stream type depends on DAI ID */ - switch (rtd->cpu_dai->id) { + switch (asoc_rtd_to_cpu(rtd, 0)->id) { case 0: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { stream_type = SST_HSW_STREAM_TYPE_SYSTEM; @@ -533,7 +533,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EINVAL; } @@ -595,7 +595,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, dmab = snd_pcm_get_dma_buf(substream); ret = create_adsp_page_table(substream, pdata, rtd, runtime->dma_area, - runtime->dma_bytes, rtd->cpu_dai->id); + runtime->dma_bytes, asoc_rtd_to_cpu(rtd, 0)->id); if (ret < 0) return ret; @@ -608,7 +608,7 @@ static int hsw_pcm_hw_params(struct snd_soc_component *component, pages = runtime->dma_bytes / PAGE_SIZE; ret = sst_hsw_stream_buffer(hsw, pcm_data->stream, - pdata->dmab[rtd->cpu_dai->id][substream->stream].addr, + pdata->dmab[asoc_rtd_to_cpu(rtd, 0)->id][substream->stream].addr, pages, runtime->dma_bytes, 0, snd_sgbuf_get_addr(dmab, 0) >> PAGE_SHIFT); if (ret < 0) { @@ -661,7 +661,7 @@ static int hsw_pcm_trigger(struct snd_soc_component *component, snd_pcm_uframes_t pos; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; sst_stream = pcm_data->stream; @@ -770,7 +770,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_soc_component *component, u32 position; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); @@ -791,7 +791,7 @@ static int hsw_pcm_open(struct snd_soc_component *component, struct sst_hsw *hsw = pdata->hsw; int dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); @@ -801,7 +801,7 @@ static int hsw_pcm_open(struct snd_soc_component *component, snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware); - pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + pcm_data->stream = sst_hsw_stream_new(hsw, asoc_rtd_to_cpu(rtd, 0)->id, hsw_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "error: failed to create stream\n"); @@ -824,7 +824,7 @@ static int hsw_pcm_close(struct snd_soc_component *component, struct sst_hsw *hsw = pdata->hsw; int ret, dai; - dai = mod_map[rtd->cpu_dai->id].dai_id; + dai = mod_map[asoc_rtd_to_cpu(rtd, 0)->id].dai_id; pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); @@ -923,9 +923,9 @@ static int hsw_pcm_new(struct snd_soc_component *component, hsw_pcm_hardware.buffer_bytes_max); } if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) - priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) - priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; + priv_data->pcm[asoc_rtd_to_cpu(rtd, 0)->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return 0; } diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..38b9d7494083 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -17,7 +17,6 @@ #include "skl.h" #define BXT_BASEFW_TIMEOUT 3000 -#define BXT_INIT_TIMEOUT 300 #define BXT_ROM_INIT_TIMEOUT 70 #define BXT_IPC_PURGE_FW 0x01004000 @@ -38,8 +37,6 @@ /* Delay before scheduling D0i3 entry */ #define BXT_D0I3_DELAY 5000 -#define BXT_FW_ROM_INIT_RETRY 3 - static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) { return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 4f64f097e9ae..c6abcd5aa67b 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -57,18 +57,34 @@ static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize) ctx->dsp_ops.stream_tag = stream_tag; memcpy(ctx->dmab.area, fwdata, fwsize); + ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK); + if (ret < 0) { + dev_err(ctx->dev, "dsp core0 power up failed\n"); + ret = -EIO; + goto base_fw_load_failed; + } + /* purge FW request */ sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR, CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE | ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID))); - ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK); + ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); if (ret < 0) { - dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret); + dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret); ret = -EIO; goto base_fw_load_failed; } + ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA, + CNL_ADSP_REG_HIPCIDA_DONE, + CNL_ADSP_REG_HIPCIDA_DONE, + BXT_INIT_TIMEOUT, "HIPCIDA Done"); + if (ret < 0) { + dev_err(ctx->dev, "timeout for purge request: %d\n", ret); + goto base_fw_load_failed; + } + /* enable interrupt */ cnl_ipc_int_enable(ctx); cnl_ipc_op_int_enable(ctx); @@ -109,7 +125,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) { struct firmware stripped_fw; struct skl_dev *cnl = ctx->thread_context; - int ret; + int ret, i; if (!ctx->fw) { ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); @@ -131,12 +147,16 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) stripped_fw.size = ctx->fw->size; skl_dsp_strip_extended_manifest(&stripped_fw); - ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); - if (ret < 0) { - dev_err(ctx->dev, "prepare firmware failed: %d\n", ret); - goto cnl_load_base_firmware_failed; + for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) { + ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); + if (!ret) + break; + dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret); } + if (ret < 0) + goto cnl_load_base_firmware_failed; + ret = sst_transfer_fw_host_dma(ctx); if (ret < 0) { dev_err(ctx->dev, "transfer firmware failed: %d\n", ret); @@ -158,6 +178,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) return 0; cnl_load_base_firmware_failed: + dev_err(ctx->dev, "firmware load failed: %d\n", ret); release_firmware(ctx->fw); ctx->fw = NULL; diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 19f328d71f24..d9c8f5cb389e 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -182,7 +182,8 @@ void skl_nhlt_remove_sysfs(struct skl_dev *skl) { struct device *dev = &skl->pci->dev; - sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); + if (skl->nhlt) + sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); } /* diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b99509675d29..89dcccdfb1cd 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -112,10 +112,7 @@ static void skl_set_suspend_active(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; struct skl_dev *skl = bus_to_skl(bus); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, substream->stream); if (w->ignore_suspend && enable) skl->supend_active++; @@ -475,10 +472,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, if (!mconfig) return -EIO; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, substream->stream); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: @@ -551,7 +545,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct skl_pipe_params p_params = {0}; struct hdac_ext_link *link; int stream_tag; @@ -650,7 +644,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; @@ -1080,7 +1074,7 @@ static int skl_platform_soc_open(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai_link = rtd->dai_link; - dev_dbg(rtd->cpu_dai->dev, "In %s:%s\n", __func__, + dev_dbg(asoc_rtd_to_cpu(rtd, 0)->dev, "In %s:%s\n", __func__, dai_link->cpus->dai_name); snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw); @@ -1232,7 +1226,7 @@ static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); u64 codec_frames, codec_nsecs; if (!codec_dai->driver->ops->delay) @@ -1287,7 +1281,7 @@ static int skl_platform_soc_get_time_info( static int skl_platform_soc_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct hdac_bus *bus = dev_get_drvdata(dai->dev); struct snd_pcm *pcm = rtd->pcm; unsigned int size; diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index cdfec0fca577..1df9ef422f61 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -67,6 +67,8 @@ struct skl_dev; #define SKL_FW_INIT 0x1 #define SKL_FW_RFW_START 0xf +#define BXT_FW_ROM_INIT_RETRY 3 +#define BXT_INIT_TIMEOUT 300 #define SKL_ADSPIC_IPC 1 #define SKL_ADSPIS_IPC 1 diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index f755ca2484cf..63182bfd7941 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -130,6 +130,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -138,6 +139,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } @@ -359,7 +361,7 @@ static int skl_resume(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct hdac_bus *bus = pci_get_drvdata(pci); struct skl_dev *skl = bus_to_skl(bus); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int ret; /* @@ -481,13 +483,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl, struct snd_soc_acpi_mach *machines) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach; - /* check if we have any codecs detected on bus */ - if (bus->codec_mask == 0) - return NULL; - /* point to common table */ mach = snd_soc_acpi_intel_hda_machines; @@ -636,6 +633,9 @@ static int skl_clock_device_register(struct skl_dev *skl) struct platform_device_info pdevinfo = {NULL}; struct skl_clk_pdata *clk_pdata; + if (!skl->nhlt) + return 0; + clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), GFP_KERNEL); if (!clk_pdata) @@ -794,7 +794,7 @@ static void skl_probe_work(struct work_struct *work) { struct skl_dev *skl = container_of(work, struct skl_dev, probe_work); struct hdac_bus *bus = skl_to_bus(skl); - struct hdac_ext_link *hlink = NULL; + struct hdac_ext_link *hlink; int err; if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { @@ -803,6 +803,9 @@ static void skl_probe_work(struct work_struct *work) return; } + skl_init_pci(skl); + skl_dum_set(bus); + err = skl_init_chip(bus, true); if (err < 0) { dev_err(bus->dev, "Init chip failed with err: %d\n", err); @@ -918,8 +921,6 @@ static int skl_first_init(struct hdac_bus *bus) return -ENXIO; } - snd_hdac_bus_reset_link(bus, true); - snd_hdac_bus_parse_capabilities(bus); /* check if PPCAP exists */ @@ -967,11 +968,7 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - /* initialize chip */ - skl_init_pci(skl); - skl_dum_set(bus); - - return skl_init_chip(bus, true); + return 0; } static int skl_probe(struct pci_dev *pci, @@ -1064,8 +1061,6 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - snd_hdac_bus_stop_chip(bus); - /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) { @@ -1082,7 +1077,8 @@ out_dsp_free: out_clk_free: skl_clock_device_unregister(skl); out_nhlt_free: - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); out_free: skl_free(bus); @@ -1131,7 +1127,8 @@ static void skl_remove(struct pci_dev *pci) skl_dmic_device_unregister(skl); skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9d5405881209..6f6f8dad0356 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -49,12 +49,8 @@ #define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 #define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 -#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24 -#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16 -#define JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_MASK \ - (0xf << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) -#define JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_MASK \ - (0x1f << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) +#define JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 24 +#define JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 16 #define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) #define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) @@ -83,16 +79,23 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf -#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_SHIFT 0 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) #define I2SDIV_IDV_SHIFT 8 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) enum jz47xx_i2s_version { JZ_I2S_JZ4740, + JZ_I2S_JZ4760, + JZ_I2S_JZ4770, JZ_I2S_JZ4780, }; +struct i2s_soc_info { + enum jz47xx_i2s_version version; + struct snd_soc_dai_driver *dai; +}; + struct jz4740_i2s { struct resource *mem; void __iomem *base; @@ -104,7 +107,7 @@ struct jz4740_i2s { struct snd_dmaengine_dai_dma_data playback_dma_data; struct snd_dmaengine_dai_dma_data capture_dma_data; - enum jz47xx_i2s_version version; + const struct i2s_soc_info *soc_info; }; static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, @@ -284,7 +287,7 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; - if (i2s->version >= JZ_I2S_JZ4780) { + if (i2s->soc_info->version >= JZ_I2S_JZ4770) { div_reg &= ~I2SDIV_IDV_MASK; div_reg |= (div - 1) << I2SDIV_IDV_SHIFT; } else { @@ -398,9 +401,9 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); - if (i2s->version >= JZ_I2S_JZ4780) { - conf = (7 << JZ4780_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | - (8 << JZ4780_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + if (i2s->soc_info->version >= JZ_I2S_JZ4760) { + conf = (7 << JZ4760_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ4760_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | JZ_AIC_CONF_OVERFLOW_PLAY_LAST | JZ_AIC_CONF_I2S | JZ_AIC_CONF_INTERNAL_CODEC; @@ -457,7 +460,17 @@ static struct snd_soc_dai_driver jz4740_i2s_dai = { .ops = &jz4740_i2s_dai_ops, }; -static struct snd_soc_dai_driver jz4780_i2s_dai = { +static const struct i2s_soc_info jz4740_i2s_soc_info = { + .version = JZ_I2S_JZ4740, + .dai = &jz4740_i2s_dai, +}; + +static const struct i2s_soc_info jz4760_i2s_soc_info = { + .version = JZ_I2S_JZ4760, + .dai = &jz4740_i2s_dai, +}; + +static struct snd_soc_dai_driver jz4770_i2s_dai = { .probe = jz4740_i2s_dai_probe, .remove = jz4740_i2s_dai_remove, .playback = { @@ -475,6 +488,16 @@ static struct snd_soc_dai_driver jz4780_i2s_dai = { .ops = &jz4740_i2s_dai_ops, }; +static const struct i2s_soc_info jz4770_i2s_soc_info = { + .version = JZ_I2S_JZ4770, + .dai = &jz4770_i2s_dai, +}; + +static const struct i2s_soc_info jz4780_i2s_soc_info = { + .version = JZ_I2S_JZ4780, + .dai = &jz4770_i2s_dai, +}; + static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", .suspend = jz4740_i2s_suspend, @@ -483,8 +506,10 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { #ifdef CONFIG_OF static const struct of_device_id jz4740_of_matches[] = { - { .compatible = "ingenic,jz4740-i2s", .data = (void *)JZ_I2S_JZ4740 }, - { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, + { .compatible = "ingenic,jz4740-i2s", .data = &jz4740_i2s_soc_info }, + { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info }, + { .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info }, + { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, jz4740_of_matches); @@ -492,45 +517,40 @@ MODULE_DEVICE_TABLE(of, jz4740_of_matches); static int jz4740_i2s_dev_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct jz4740_i2s *i2s; struct resource *mem; int ret; - i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->version = - (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev); + i2s->soc_info = device_get_match_data(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - i2s->base = devm_ioremap_resource(&pdev->dev, mem); + i2s->base = devm_ioremap_resource(dev, mem); if (IS_ERR(i2s->base)) return PTR_ERR(i2s->base); i2s->phys_base = mem->start; - i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + i2s->clk_aic = devm_clk_get(dev, "aic"); if (IS_ERR(i2s->clk_aic)) return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + i2s->clk_i2s = devm_clk_get(dev, "i2s"); if (IS_ERR(i2s->clk_i2s)) return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - if (i2s->version == JZ_I2S_JZ4780) - ret = devm_snd_soc_register_component(&pdev->dev, - &jz4740_i2s_component, &jz4780_i2s_dai, 1); - else - ret = devm_snd_soc_register_component(&pdev->dev, - &jz4740_i2s_component, &jz4740_i2s_dai, 1); - + ret = devm_snd_soc_register_component(dev, &jz4740_i2s_component, + i2s->soc_info->dai, 1); if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + return devm_snd_dmaengine_pcm_register(dev, NULL, SND_DMAENGINE_PCM_FLAG_COMPAT); } diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index 8c3c808bda9a..4f66b011f1b4 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -19,7 +19,7 @@ static int a370db_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int freq; switch (params_rate(params)) { diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index f882b4003edf..e037826b2451 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -20,7 +20,7 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; - return snd_soc_dai_get_drvdata(soc_runtime->cpu_dai); + return snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(soc_runtime, 0)); } static const struct snd_pcm_hardware kirkwood_dma_snd_hw = { diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 4254f3a954dd..375e3b492922 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -40,7 +40,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct snd_pcm_runtime *runtime = substream->runtime; - int memif_num = rtd->cpu_dai->id; + int memif_num = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[memif_num]; const struct snd_pcm_hardware *mtk_afe_hardware = afe->mtk_afe_hardware; int ret; @@ -100,7 +100,7 @@ void mtk_afe_fe_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; int irq_id; irq_id = memif->irq_usage; @@ -122,7 +122,7 @@ int mtk_afe_fe_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; int ret; unsigned int channels = params_channels(params); @@ -199,7 +199,7 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime * const runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; struct mtk_base_afe_irq *irqs = &afe->irqs[memif->irq_usage]; const struct mtk_base_irq_data *irq_data = irqs->irq_data; @@ -265,7 +265,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; int pbuf_size; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 44dfef713905..0a1a65c86f0e 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -82,7 +82,7 @@ snd_pcm_uframes_t mtk_afe_pcm_pointer(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; const struct mtk_base_memif_data *memif_data = memif->data; struct regmap *regmap = afe->regmap; struct device *dev = afe->dev; diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 488603a0c4b1..f0250b0dd734 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -497,7 +497,7 @@ static int mt2701_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; int fs; - if (rtd->cpu_dai->id != MT2701_MEMIF_ULBT) + if (asoc_rtd_to_cpu(rtd, 0)->id != MT2701_MEMIF_ULBT) fs = mt2701_afe_i2s_fs(rate); else fs = (rate == 16000 ? 1 : 0); diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index b6941796efca..c47af9b6949b 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -128,8 +128,8 @@ static int mt2701_cs42448_be_ops_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int mclk_rate; unsigned int rate = params_rate(params); unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4; diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 8c4c89e4c616..0122e7df067f 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -25,8 +25,8 @@ static int mt2701_wm8960_be_ops_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int mclk_rate; unsigned int rate = params_rate(params); unsigned int div_mclk_over_bck = rate > 192000 ? 2 : 4; diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index 378bfc16ef52..7f930556d961 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -143,7 +143,7 @@ static int mt6797_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; return mt6797_rate_transform(afe->dev, rate, id); } diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 461e4de8c918..1e3f2d786066 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -485,7 +485,7 @@ static int mt8173_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; + struct mtk_base_afe_memif *memif = &afe->memif[asoc_rtd_to_cpu(rtd, 0)->id]; int fs; if (memif->data->id == MT8173_AFE_MEMIF_DAI || diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 22c00600c999..37693d354e66 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -53,7 +53,7 @@ static int mt8173_max98090_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256, SND_SOC_CLOCK_IN); @@ -67,7 +67,7 @@ static int mt8173_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 2e1e61d8f127..51009a172777 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -47,7 +47,7 @@ static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); @@ -73,7 +73,7 @@ static struct snd_soc_jack mt8173_rt5650_rt5514_jack; static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int ret; rt5645_sel_asrc_clk_src(component, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index ebcc0b86286b..247ac7690805 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -51,7 +51,7 @@ static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk 12.288M */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, params_rate(params) * 512); @@ -77,8 +77,8 @@ static struct snd_soc_jack mt8173_rt5650_rt5676_jack; static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; - struct snd_soc_component *component_sub = runtime->codec_dais[1]->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_component *component_sub = asoc_rtd_to_codec(runtime, 1)->component; int ret; rt5645_sel_asrc_clk_src(component, diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef6f23675286..2065c94dbf99 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -11,6 +11,7 @@ #include <linux/of_gpio.h> #include <sound/soc.h> #include <sound/jack.h> +#include <sound/hdmi-codec.h> #include "../../codecs/rt5645.h" #define MCLK_FOR_CODECS 12288000 @@ -77,7 +78,7 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, break; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* pll from mclk */ ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock, params_rate(params) * 512); @@ -98,13 +99,13 @@ static const struct snd_soc_ops mt8173_rt5650_ops = { .hw_params = mt8173_rt5650_hw_params, }; -static struct snd_soc_jack mt8173_rt5650_jack; +static struct snd_soc_jack mt8173_rt5650_jack, mt8173_rt5650_hdmi_jack; static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dais[0]->component; - const char *codec_capture_dai = runtime->codec_dais[1]->name; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + const char *codec_capture_dai = asoc_rtd_to_codec(runtime, 1)->name; int ret; rt5645_sel_asrc_clk_src(component, @@ -144,6 +145,19 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) &mt8173_rt5650_jack); } +static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, + &mt8173_rt5650_hdmi_jack, NULL, 0); + if (ret) + return ret; + + return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component, + &mt8173_rt5650_hdmi_jack); +} + enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, @@ -222,6 +236,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .name = "HDMI BE", .no_pcm = 1, .dpcm_playback = 1, + .init = mt8173_rt5650_hdmi_init, SND_SOC_DAILINK_REG(hdmi_be), }, }; diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 6e2270bbb10e..c8ded53bde1d 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -146,7 +146,7 @@ static int mt8183_memif_fs(struct snd_pcm_substream *substream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int id = rtd->cpu_dai->id; + int id = asoc_rtd_to_cpu(rtd, 0)->id; return mt8183_rate_transform(afe->dev, rate, id); } diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c65493721e90..5b3dfa79b4ae 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -16,7 +16,9 @@ #include "../../codecs/da7219-aad.h" #include "../../codecs/da7219.h" -static struct snd_soc_jack headset_jack; +struct mt8183_da7219_max98357_priv { + struct snd_soc_jack headset_jack; +}; static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -26,7 +28,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int mclk_fs_ratio = 128; unsigned int mclk_fs = rate * mclk_fs_ratio; - return snd_soc_dai_set_sysclk(rtd->cpu_dai, + return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); } @@ -38,19 +40,19 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; unsigned int rate = params_rate(params); unsigned int mclk_fs_ratio = 256; unsigned int mclk_fs = rate * mclk_fs_ratio; unsigned int freq; int ret = 0, j; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_sysclk(codec_dai, @@ -80,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_pll(codec_dai, @@ -116,6 +118,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static int +mt8183_da7219_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = { + .startup = mt8183_da7219_max98357_bt_sco_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -222,6 +264,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(playback2), }, { @@ -240,6 +283,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(capture1), }, { @@ -351,8 +395,12 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { { .name = "TDM", .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_IF | + SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, SND_SOC_DAILINK_REG(tdm), }, }; @@ -372,9 +420,31 @@ static struct snd_soc_codec_conf mt6358_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_max98357_dapm_routes[] = { + {"Speakers", NULL, "Speaker"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_max98357_card = { .name = "mt8183_da7219_max98357", .owner = THIS_MODULE, + .controls = mt8183_da7219_max98357_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), + .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), + .dapm_routes = mt8183_da7219_max98357_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), .dai_link = mt8183_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_max98357_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, @@ -387,6 +457,8 @@ static int mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) { int ret; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(component->card); /* Enable Headset and 4 Buttons Jack detection */ ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card, @@ -394,12 +466,12 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &headset_jack, + &priv->headset_jack, NULL, 0); if (ret) return ret; - da7219_aad_jack_det(component, &headset_jack); + da7219_aad_jack_det(component, &priv->headset_jack); return ret; } @@ -409,7 +481,8 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; + struct pinctrl *pinctrl; int ret, i; card->dev = &pdev->dev; @@ -436,22 +509,21 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + pinctrl = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); + if (IS_ERR(pinctrl)) { + ret = PTR_ERR(pinctrl); + dev_err(&pdev->dev, "%s failed to select default state %d\n", __func__, ret); return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", - __func__); - return PTR_ERR(default_pins); - } - - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } #ifdef CONFIG_OF @@ -478,4 +550,3 @@ MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver"); MODULE_AUTHOR("Shunli Wang <shunli.wang@mediatek.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("mt8183_da7219_max98357 soc card"); - diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 0555f7d73d05..1fca8df109b4 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -41,7 +41,7 @@ static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int mclk_fs_ratio = 128; unsigned int mclk_fs = rate * mclk_fs_ratio; - return snd_soc_dai_set_sysclk(rtd->cpu_dai, + return snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk_fs, SND_SOC_CLOCK_OUT); } diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 2e3676147cea..8b6295283989 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -2,6 +2,16 @@ menu "ASoC support for Amlogic platforms" depends on ARCH_MESON || COMPILE_TEST +config SND_MESON_AIU + tristate "Amlogic AIU" + select SND_MESON_CODEC_GLUE + select SND_PCM_IEC958 + imply SND_SOC_MESON_T9015 + imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI + help + Select Y or M to add support for the Audio output subsystem found + in the Amlogic Meson8, Meson8b and GX SoC families + config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO @@ -50,6 +60,7 @@ config SND_MESON_AXG_TDMOUT config SND_MESON_AXG_SOUND_CARD tristate "Amlogic AXG Sound Card Support" select SND_MESON_AXG_TDM_INTERFACE + select SND_MESON_CARD_UTILS imply SND_MESON_AXG_FRDDR imply SND_MESON_AXG_TODDR imply SND_MESON_AXG_TDMIN @@ -85,11 +96,41 @@ config SND_MESON_AXG_PDM Select Y or M to add support for PDM input embedded in the Amlogic AXG SoC family +config SND_MESON_CARD_UTILS + tristate + +config SND_MESON_CODEC_GLUE + tristate + +config SND_MESON_GX_SOUND_CARD + tristate "Amlogic GX Sound Card Support" + select SND_MESON_CARD_UTILS + imply SND_MESON_AIU + help + Select Y or M to add support for the GXBB/GXL SoC sound card + +config SND_MESON_G12A_TOACODEC + tristate "Amlogic G12A To Internal DAC Control Support" + select SND_MESON_CODEC_GLUE + select REGMAP_MMIO + imply SND_SOC_MESON_T9015 + help + Select Y or M to add support for the internal audio DAC on the + g12a SoC family + config SND_MESON_G12A_TOHDMITX tristate "Amlogic G12A To HDMI TX Control Support" select REGMAP_MMIO + select SND_MESON_CODEC_GLUE imply SND_SOC_HDMI_CODEC help Select Y or M to add support for HDMI audio on the g12a SoC family + +config SND_SOC_MESON_T9015 + tristate "Amlogic T9015 DAC" + select REGMAP_MMIO + help + Say Y or M if you want to add support for the internal DAC found + on GXL, G12 and SM1 SoC family. endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index 1a8b1470ed84..e446bc980481 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,5 +1,13 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) +snd-soc-meson-aiu-objs := aiu.o +snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o +snd-soc-meson-aiu-objs += aiu-codec-ctrl.o +snd-soc-meson-aiu-objs += aiu-encoder-i2s.o +snd-soc-meson-aiu-objs += aiu-encoder-spdif.o +snd-soc-meson-aiu-objs += aiu-fifo.o +snd-soc-meson-aiu-objs += aiu-fifo-i2s.o +snd-soc-meson-aiu-objs += aiu-fifo-spdif.o snd-soc-meson-axg-fifo-objs := axg-fifo.o snd-soc-meson-axg-frddr-objs := axg-frddr.o snd-soc-meson-axg-toddr-objs := axg-toddr.o @@ -11,8 +19,14 @@ snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifin-objs := axg-spdifin.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o snd-soc-meson-axg-pdm-objs := axg-pdm.o +snd-soc-meson-card-utils-objs := meson-card-utils.o +snd-soc-meson-codec-glue-objs := meson-codec-glue.o +snd-soc-meson-gx-sound-card-objs := gx-card.o +snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o +snd-soc-meson-t9015-objs := t9015.o +obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o obj-$(CONFIG_SND_MESON_AXG_TODDR) += snd-soc-meson-axg-toddr.o @@ -24,4 +38,9 @@ obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFIN) += snd-soc-meson-axg-spdifin.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o +obj-$(CONFIG_SND_MESON_CARD_UTILS) += snd-soc-meson-card-utils.o +obj-$(CONFIG_SND_MESON_CODEC_GLUE) += snd-soc-meson-codec-glue.o +obj-$(CONFIG_SND_MESON_GX_SOUND_CARD) += snd-soc-meson-gx-sound-card.o +obj-$(CONFIG_SND_MESON_G12A_TOACODEC) += snd-soc-meson-g12a-toacodec.o obj-$(CONFIG_SND_MESON_G12A_TOHDMITX) += snd-soc-meson-g12a-tohdmitx.o +obj-$(CONFIG_SND_SOC_MESON_T9015) += snd-soc-meson-t9015.o diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c new file mode 100644 index 000000000000..7078197e0cc5 --- /dev/null +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -0,0 +1,203 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <dt-bindings/sound/meson-aiu.h> +#include "aiu.h" +#include "meson-codec-glue.h" + +#define CTRL_DIN_EN 15 +#define CTRL_CLK_INV BIT(14) +#define CTRL_LRCLK_INV BIT(13) +#define CTRL_I2S_IN_BCLK_SRC BIT(11) +#define CTRL_DIN_LRCLK_SRC_SHIFT 6 +#define CTRL_DIN_LRCLK_SRC (0x3 << CTRL_DIN_LRCLK_SRC_SHIFT) +#define CTRL_BCLK_MCLK_SRC GENMASK(5, 4) +#define CTRL_DIN_SKEW GENMASK(3, 2) +#define CTRL_I2S_OUT_LANE_SRC 0 + +#define AIU_ACODEC_OUT_CHMAX 2 + +static const char * const aiu_acodec_ctrl_mux_texts[] = { + "DISABLED", "I2S", "PCM", +}; + +static int aiu_acodec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL_DIN_LRCLK_SRC, + FIELD_PREP(CTRL_DIN_LRCLK_SRC, + mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL_DIN_LRCLK_SRC | + CTRL_BCLK_MCLK_SRC, + FIELD_PREP(CTRL_DIN_LRCLK_SRC, mux) | + FIELD_PREP(CTRL_BCLK_MCLK_SRC, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(aiu_acodec_ctrl_mux_enum, AIU_ACODEC_CTRL, + CTRL_DIN_LRCLK_SRC_SHIFT, + aiu_acodec_ctrl_mux_texts); + +static const struct snd_kcontrol_new aiu_acodec_ctrl_mux = + SOC_DAPM_ENUM_EXT("ACodec Source", aiu_acodec_ctrl_mux_enum, + snd_soc_dapm_get_enum_double, + aiu_acodec_ctrl_mux_put_enum); + +static const struct snd_kcontrol_new aiu_acodec_ctrl_out_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", AIU_ACODEC_CTRL, + CTRL_DIN_EN, 1, 0); + +static const struct snd_soc_dapm_widget aiu_acodec_ctrl_widgets[] = { + SND_SOC_DAPM_MUX("ACODEC SRC", SND_SOC_NOPM, 0, 0, + &aiu_acodec_ctrl_mux), + SND_SOC_DAPM_SWITCH("ACODEC OUT EN", SND_SOC_NOPM, 0, 0, + &aiu_acodec_ctrl_out_enable), +}; + +static int aiu_acodec_ctrl_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + int ret; + + ret = meson_codec_glue_input_hw_params(substream, params, dai); + if (ret) + return ret; + + /* The glue will provide 1 lane out of the 4 to the output */ + data = meson_codec_glue_input_get_data(dai); + data->params.channels_min = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX, + data->params.channels_min); + data->params.channels_max = min_t(unsigned int, AIU_ACODEC_OUT_CHMAX, + data->params.channels_max); + + return 0; +} + +static const struct snd_soc_dai_ops aiu_acodec_ctrl_input_ops = { + .hw_params = aiu_acodec_ctrl_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops aiu_acodec_ctrl_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define AIU_ACODEC_CTRL_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AIU_ACODEC_STREAM(xname, xsuffix, xchmax) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = (xchmax), \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AIU_ACODEC_CTRL_FORMATS, \ +} + +#define AIU_ACODEC_INPUT(xname) { \ + .name = "ACODEC CTRL " xname, \ + .playback = AIU_ACODEC_STREAM(xname, "Playback", 8), \ + .ops = &aiu_acodec_ctrl_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define AIU_ACODEC_OUTPUT(xname) { \ + .name = "ACODEC CTRL " xname, \ + .capture = AIU_ACODEC_STREAM(xname, "Capture", AIU_ACODEC_OUT_CHMAX), \ + .ops = &aiu_acodec_ctrl_output_ops, \ +} + +static struct snd_soc_dai_driver aiu_acodec_ctrl_dai_drv[] = { + [CTRL_I2S] = AIU_ACODEC_INPUT("ACODEC I2S IN"), + [CTRL_PCM] = AIU_ACODEC_INPUT("ACODEC PCM IN"), + [CTRL_OUT] = AIU_ACODEC_OUTPUT("ACODEC OUT"), +}; + +static const struct snd_soc_dapm_route aiu_acodec_ctrl_routes[] = { + { "ACODEC SRC", "I2S", "ACODEC I2S IN Playback" }, + { "ACODEC SRC", "PCM", "ACODEC PCM IN Playback" }, + { "ACODEC OUT EN", "Switch", "ACODEC SRC" }, + { "ACODEC OUT Capture", NULL, "ACODEC OUT EN" }, +}; + +static const struct snd_kcontrol_new aiu_acodec_ctrl_controls[] = { + SOC_SINGLE("ACODEC I2S Lane Select", AIU_ACODEC_CTRL, + CTRL_I2S_OUT_LANE_SRC, 3, 0), +}; + +static int aiu_acodec_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_ACODEC); +} + +static int aiu_acodec_ctrl_component_probe(struct snd_soc_component *component) +{ + /* + * NOTE: Din Skew setting + * According to the documentation, the following update adds one delay + * to the din line. Without this, the output saturates. This happens + * regardless of the link format (i2s or left_j) so it is not clear what + * it actually does but it seems to be required + */ + snd_soc_component_update_bits(component, AIU_ACODEC_CTRL, + CTRL_DIN_SKEW, + FIELD_PREP(CTRL_DIN_SKEW, 2)); + + return 0; +} + +static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { + .name = "AIU Internal DAC Codec Control", + .probe = aiu_acodec_ctrl_component_probe, + .controls = aiu_acodec_ctrl_controls, + .num_controls = ARRAY_SIZE(aiu_acodec_ctrl_controls), + .dapm_widgets = aiu_acodec_ctrl_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_acodec_ctrl_widgets), + .dapm_routes = aiu_acodec_ctrl_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_acodec_ctrl_routes), + .of_xlate_dai_name = aiu_acodec_of_xlate_dai_name, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +int aiu_acodec_ctrl_register_component(struct device *dev) +{ + return snd_soc_register_component(dev, &aiu_acodec_ctrl_component, + aiu_acodec_ctrl_dai_drv, + ARRAY_SIZE(aiu_acodec_ctrl_dai_drv)); +} diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c new file mode 100644 index 000000000000..4b773d3e8b07 --- /dev/null +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -0,0 +1,151 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <dt-bindings/sound/meson-aiu.h> +#include "aiu.h" +#include "meson-codec-glue.h" + +#define CTRL_CLK_SEL GENMASK(1, 0) +#define CTRL_DATA_SEL_SHIFT 4 +#define CTRL_DATA_SEL (0x3 << CTRL_DATA_SEL_SHIFT) + +static const char * const aiu_codec_ctrl_mux_texts[] = { + "DISABLED", "PCM", "I2S", +}; + +static int aiu_codec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL_DATA_SEL, + FIELD_PREP(CTRL_DATA_SEL, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + /* Reset the source first */ + snd_soc_component_update_bits(component, e->reg, + CTRL_CLK_SEL | + CTRL_DATA_SEL, + FIELD_PREP(CTRL_CLK_SEL, 0) | + FIELD_PREP(CTRL_DATA_SEL, 0)); + + /* Set the appropriate source */ + snd_soc_component_update_bits(component, e->reg, + CTRL_CLK_SEL | + CTRL_DATA_SEL, + FIELD_PREP(CTRL_CLK_SEL, mux) | + FIELD_PREP(CTRL_DATA_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(aiu_hdmi_ctrl_mux_enum, AIU_HDMI_CLK_DATA_CTRL, + CTRL_DATA_SEL_SHIFT, + aiu_codec_ctrl_mux_texts); + +static const struct snd_kcontrol_new aiu_hdmi_ctrl_mux = + SOC_DAPM_ENUM_EXT("HDMI Source", aiu_hdmi_ctrl_mux_enum, + snd_soc_dapm_get_enum_double, + aiu_codec_ctrl_mux_put_enum); + +static const struct snd_soc_dapm_widget aiu_hdmi_ctrl_widgets[] = { + SND_SOC_DAPM_MUX("HDMI CTRL SRC", SND_SOC_NOPM, 0, 0, + &aiu_hdmi_ctrl_mux), +}; + +static const struct snd_soc_dai_ops aiu_codec_ctrl_input_ops = { + .hw_params = meson_codec_glue_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops aiu_codec_ctrl_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define AIU_CODEC_CTRL_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define AIU_CODEC_CTRL_STREAM(xname, xsuffix) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = 8, \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AIU_CODEC_CTRL_FORMATS, \ +} + +#define AIU_CODEC_CTRL_INPUT(xname) { \ + .name = "CODEC CTRL " xname, \ + .playback = AIU_CODEC_CTRL_STREAM(xname, "Playback"), \ + .ops = &aiu_codec_ctrl_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define AIU_CODEC_CTRL_OUTPUT(xname) { \ + .name = "CODEC CTRL " xname, \ + .capture = AIU_CODEC_CTRL_STREAM(xname, "Capture"), \ + .ops = &aiu_codec_ctrl_output_ops, \ +} + +static struct snd_soc_dai_driver aiu_hdmi_ctrl_dai_drv[] = { + [CTRL_I2S] = AIU_CODEC_CTRL_INPUT("HDMI I2S IN"), + [CTRL_PCM] = AIU_CODEC_CTRL_INPUT("HDMI PCM IN"), + [CTRL_OUT] = AIU_CODEC_CTRL_OUTPUT("HDMI OUT"), +}; + +static const struct snd_soc_dapm_route aiu_hdmi_ctrl_routes[] = { + { "HDMI CTRL SRC", "I2S", "HDMI I2S IN Playback" }, + { "HDMI CTRL SRC", "PCM", "HDMI PCM IN Playback" }, + { "HDMI OUT Capture", NULL, "HDMI CTRL SRC" }, +}; + +static int aiu_hdmi_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_HDMI); +} + +static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { + .name = "AIU HDMI Codec Control", + .dapm_widgets = aiu_hdmi_ctrl_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_hdmi_ctrl_widgets), + .dapm_routes = aiu_hdmi_ctrl_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_hdmi_ctrl_routes), + .of_xlate_dai_name = aiu_hdmi_of_xlate_dai_name, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +int aiu_hdmi_ctrl_register_component(struct device *dev) +{ + return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component, + aiu_hdmi_ctrl_dai_drv, + ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv)); +} + diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c new file mode 100644 index 000000000000..832e22d275fe --- /dev/null +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -0,0 +1,365 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "aiu.h" + +#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0) +#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5) +#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9) +#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11) +#define AIU_RST_SOFT_I2S_FAST BIT(0) + +#define AIU_I2S_DAC_CFG_MSB_FIRST BIT(2) +#define AIU_I2S_MISC_HOLD_EN BIT(2) +#define AIU_CLK_CTRL_I2S_DIV_EN BIT(0) +#define AIU_CLK_CTRL_I2S_DIV GENMASK(3, 2) +#define AIU_CLK_CTRL_AOCLK_INVERT BIT(6) +#define AIU_CLK_CTRL_LRCLK_INVERT BIT(7) +#define AIU_CLK_CTRL_LRCLK_SKEW GENMASK(9, 8) +#define AIU_CLK_CTRL_MORE_HDMI_AMCLK BIT(6) +#define AIU_CLK_CTRL_MORE_I2S_DIV GENMASK(5, 0) +#define AIU_CODEC_DAC_LRCLK_CTRL_DIV GENMASK(11, 0) + +static void aiu_encoder_i2s_divider_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV_EN, + enable ? AIU_CLK_CTRL_I2S_DIV_EN : 0); +} + +static void aiu_encoder_i2s_hold(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_I2S_MISC, + AIU_I2S_MISC_HOLD_EN, + enable ? AIU_I2S_MISC_HOLD_EN : 0); +} + +static int aiu_encoder_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_encoder_i2s_hold(component, false); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + aiu_encoder_i2s_hold(component, true); + return 0; + + default: + return -EINVAL; + } +} + +static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + /* Always operate in split (classic interleaved) mode */ + unsigned int desc = AIU_I2S_SOURCE_DESC_MODE_SPLIT; + unsigned int val; + + /* Reset required to update the pipeline */ + snd_soc_component_write(component, AIU_RST_SOFT, AIU_RST_SOFT_I2S_FAST); + snd_soc_component_read(component, AIU_I2S_SYNC, &val); + + switch (params_physical_width(params)) { + case 16: /* Nothing to do */ + break; + + case 32: + desc |= (AIU_I2S_SOURCE_DESC_MODE_24BIT | + AIU_I2S_SOURCE_DESC_MODE_32BIT); + break; + + default: + return -EINVAL; + } + + switch (params_channels(params)) { + case 2: /* Nothing to do */ + break; + case 8: + desc |= AIU_I2S_SOURCE_DESC_MODE_8CH; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_I2S_SOURCE_DESC, + AIU_I2S_SOURCE_DESC_MODE_8CH | + AIU_I2S_SOURCE_DESC_MODE_24BIT | + AIU_I2S_SOURCE_DESC_MODE_32BIT | + AIU_I2S_SOURCE_DESC_MODE_SPLIT, + desc); + + return 0; +} + +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) +{ + switch (bs) { + case 1: + case 2: + case 4: + case 8: + /* These are the only valid legacy dividers */ + break; + + default: + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, + __ffs(bs))); + + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, + 0)); + + return 0; +} + +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) +{ + /* + * NOTE: this HW is odd. + * In most configuration, the i2s divider is 'mclk / blck'. + * However, in 16 bits - 8ch mode, this factor needs to be + * increased by 50% to get the correct output rate. + * No idea why ! + */ + if (params_width(params) == 16 && params_channels(params) == 8) { + if (bs % 2) { + dev_err(component->dev, + "Cannot increase i2s divider by 50%%\n"); + return -EINVAL; + } + bs += bs / 2; + } + + /* Use CLK_MORE for mclk to bclk divider */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); + + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, + bs - 1)); + + return 0; +} + +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int srate = params_rate(params); + unsigned int fs, bs; + int ret; + + /* Get the oversampling factor */ + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + + if (fs % 64) + return -EINVAL; + + /* Send data MSB first */ + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, + AIU_I2S_DAC_CFG_MSB_FIRST, + AIU_I2S_DAC_CFG_MSB_FIRST); + + /* Set bclk to lrlck ratio */ + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, + AIU_CODEC_DAC_LRCLK_CTRL_DIV, + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, + 64 - 1)); + + bs = fs / 64; + + if (aiu->platform->has_clk_ctrl_more_i2s_div) + ret = aiu_encoder_i2s_set_more_div(component, params, bs); + else + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); + + if (ret) + return ret; + + /* Make sure amclk is used for HDMI i2s as well */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_HDMI_AMCLK, + AIU_CLK_CTRL_MORE_HDMI_AMCLK); + + return 0; +} + +static int aiu_encoder_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + /* Disable the clock while changing the settings */ + aiu_encoder_i2s_divider_enable(component, false); + + ret = aiu_encoder_i2s_setup_desc(component, params); + if (ret) { + dev_err(dai->dev, "setting i2s desc failed\n"); + return ret; + } + + ret = aiu_encoder_i2s_set_clocks(component, params); + if (ret) { + dev_err(dai->dev, "setting i2s clocks failed\n"); + return ret; + } + + aiu_encoder_i2s_divider_enable(component, true); + + return 0; +} + +static int aiu_encoder_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + aiu_encoder_i2s_divider_enable(component, false); + + return 0; +} + +static int aiu_encoder_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int inv = fmt & SND_SOC_DAIFMT_INV_MASK; + unsigned int val = 0; + unsigned int skew; + + /* Only CPU Master / Codec Slave supported ATM */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + if (inv == SND_SOC_DAIFMT_NB_IF || + inv == SND_SOC_DAIFMT_IB_IF) + val |= AIU_CLK_CTRL_LRCLK_INVERT; + + if (inv == SND_SOC_DAIFMT_IB_NF || + inv == SND_SOC_DAIFMT_IB_IF) + val |= AIU_CLK_CTRL_AOCLK_INVERT; + + /* Signal skew */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Invert sample clock for i2s */ + val ^= AIU_CLK_CTRL_LRCLK_INVERT; + skew = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + skew = 0; + break; + default: + return -EINVAL; + } + + val |= FIELD_PREP(AIU_CLK_CTRL_LRCLK_SKEW, skew); + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_LRCLK_INVERT | + AIU_CLK_CTRL_AOCLK_INVERT | + AIU_CLK_CTRL_LRCLK_SKEW, + val); + + return 0; +} + +static int aiu_encoder_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + if (WARN_ON(clk_id != 0)) + return -EINVAL; + + if (dir == SND_SOC_CLOCK_IN) + return 0; + + ret = clk_set_rate(aiu->i2s.clks[MCLK].clk, freq); + if (ret) + dev_err(dai->dev, "Failed to set sysclk to %uHz", freq); + + return ret; +} + +static const unsigned int hw_channels[] = {2, 8}; +static const struct snd_pcm_hw_constraint_list hw_channel_constraints = { + .list = hw_channels, + .count = ARRAY_SIZE(hw_channels), + .mask = 0, +}; + +static int aiu_encoder_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + /* Make sure the encoder gets either 2 or 8 channels */ + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_channel_constraints); + if (ret) { + dev_err(dai->dev, "adding channels constraints failed\n"); + return ret; + } + + ret = clk_bulk_prepare_enable(aiu->i2s.clk_num, aiu->i2s.clks); + if (ret) + dev_err(dai->dev, "failed to enable i2s clocks\n"); + + return ret; +} + +static void aiu_encoder_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + + clk_bulk_disable_unprepare(aiu->i2s.clk_num, aiu->i2s.clks); +} + +const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops = { + .trigger = aiu_encoder_i2s_trigger, + .hw_params = aiu_encoder_i2s_hw_params, + .hw_free = aiu_encoder_i2s_hw_free, + .set_fmt = aiu_encoder_i2s_set_fmt, + .set_sysclk = aiu_encoder_i2s_set_sysclk, + .startup = aiu_encoder_i2s_startup, + .shutdown = aiu_encoder_i2s_shutdown, +}; + diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c new file mode 100644 index 000000000000..de850913975f --- /dev/null +++ b/sound/soc/meson/aiu-encoder-spdif.c @@ -0,0 +1,209 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <sound/pcm_params.h> +#include <sound/pcm_iec958.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "aiu.h" + +#define AIU_958_MISC_NON_PCM BIT(0) +#define AIU_958_MISC_MODE_16BITS BIT(1) +#define AIU_958_MISC_16BITS_ALIGN GENMASK(6, 5) +#define AIU_958_MISC_MODE_32BITS BIT(7) +#define AIU_958_MISC_U_FROM_STREAM BIT(12) +#define AIU_958_MISC_FORCE_LR BIT(13) +#define AIU_958_CTRL_HOLD_EN BIT(0) +#define AIU_CLK_CTRL_958_DIV_EN BIT(1) +#define AIU_CLK_CTRL_958_DIV GENMASK(5, 4) +#define AIU_CLK_CTRL_958_DIV_MORE BIT(12) + +#define AIU_CS_WORD_LEN 4 +#define AIU_958_INTERNAL_DIV 2 + +static void +aiu_encoder_spdif_divider_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_958_DIV_EN, + enable ? AIU_CLK_CTRL_958_DIV_EN : 0); +} + +static void aiu_encoder_spdif_hold(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_958_CTRL, + AIU_958_CTRL_HOLD_EN, + enable ? AIU_958_CTRL_HOLD_EN : 0); +} + +static int +aiu_encoder_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_encoder_spdif_hold(component, false); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + aiu_encoder_spdif_hold(component, true); + return 0; + + default: + return -EINVAL; + } +} + +static int aiu_encoder_spdif_setup_cs_word(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + u8 cs[AIU_CS_WORD_LEN]; + unsigned int val; + int ret; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, cs, + AIU_CS_WORD_LEN); + if (ret < 0) + return ret; + + /* Write the 1st half word */ + val = cs[1] | cs[0] << 8; + snd_soc_component_write(component, AIU_958_CHSTAT_L0, val); + snd_soc_component_write(component, AIU_958_CHSTAT_R0, val); + + /* Write the 2nd half word */ + val = cs[3] | cs[2] << 8; + snd_soc_component_write(component, AIU_958_CHSTAT_L1, val); + snd_soc_component_write(component, AIU_958_CHSTAT_R1, val); + + return 0; +} + +static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mrate; + int ret; + + /* Disable the clock while changing the settings */ + aiu_encoder_spdif_divider_enable(component, false); + + switch (params_physical_width(params)) { + case 16: + val |= AIU_958_MISC_MODE_16BITS; + val |= FIELD_PREP(AIU_958_MISC_16BITS_ALIGN, 2); + break; + case 32: + val |= AIU_958_MISC_MODE_32BITS; + break; + default: + dev_err(dai->dev, "Unsupport physical width\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_958_MISC, + AIU_958_MISC_NON_PCM | + AIU_958_MISC_MODE_16BITS | + AIU_958_MISC_16BITS_ALIGN | + AIU_958_MISC_MODE_32BITS | + AIU_958_MISC_FORCE_LR | + AIU_958_MISC_U_FROM_STREAM, + val); + + /* Set the stream channel status word */ + ret = aiu_encoder_spdif_setup_cs_word(component, params); + if (ret) { + dev_err(dai->dev, "failed to set channel status word\n"); + return ret; + } + + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_958_DIV | + AIU_CLK_CTRL_958_DIV_MORE, + FIELD_PREP(AIU_CLK_CTRL_958_DIV, + __ffs(AIU_958_INTERNAL_DIV))); + + /* 2 * 32bits per subframe * 2 channels = 128 */ + mrate = params_rate(params) * 128 * AIU_958_INTERNAL_DIV; + ret = clk_set_rate(aiu->spdif.clks[MCLK].clk, mrate); + if (ret) { + dev_err(dai->dev, "failed to set mclk rate\n"); + return ret; + } + + aiu_encoder_spdif_divider_enable(component, true); + + return 0; +} + +static int aiu_encoder_spdif_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + aiu_encoder_spdif_divider_enable(component, false); + + return 0; +} + +static int aiu_encoder_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + int ret; + + /* + * NOTE: Make sure the spdif block is on its own divider. + * + * The spdif can be clocked by the i2s master clock or its own + * clock. We should (in theory) change the source depending on the + * origin of the data. + * + * However, considering the clocking scheme used on these platforms, + * the master clocks will pick the same PLL source when they are + * playing from the same FIFO. The clock should be in sync so, it + * should not be necessary to reparent the spdif master clock. + */ + ret = clk_set_parent(aiu->spdif.clks[MCLK].clk, + aiu->spdif_mclk); + if (ret) + return ret; + + ret = clk_bulk_prepare_enable(aiu->spdif.clk_num, aiu->spdif.clks); + if (ret) + dev_err(dai->dev, "failed to enable spdif clocks\n"); + + return ret; +} + +static void aiu_encoder_spdif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(dai->component); + + clk_bulk_disable_unprepare(aiu->spdif.clk_num, aiu->spdif.clks); +} + +const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops = { + .trigger = aiu_encoder_spdif_trigger, + .hw_params = aiu_encoder_spdif_hw_params, + .hw_free = aiu_encoder_spdif_hw_free, + .startup = aiu_encoder_spdif_startup, + .shutdown = aiu_encoder_spdif_shutdown, +}; diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c new file mode 100644 index 000000000000..9a5271ce80fe --- /dev/null +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -0,0 +1,153 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_I2S_SOURCE_DESC_MODE_8CH BIT(0) +#define AIU_I2S_SOURCE_DESC_MODE_24BIT BIT(5) +#define AIU_I2S_SOURCE_DESC_MODE_32BIT BIT(9) +#define AIU_I2S_SOURCE_DESC_MODE_SPLIT BIT(11) +#define AIU_MEM_I2S_MASKS_IRQ_BLOCK GENMASK(31, 16) +#define AIU_MEM_I2S_CONTROL_MODE_16BIT BIT(6) +#define AIU_MEM_I2S_BUF_CNTL_INIT BIT(0) +#define AIU_RST_SOFT_I2S_FAST BIT(0) + +#define AIU_FIFO_I2S_BLOCK 256 + +static struct snd_pcm_hardware fifo_i2s_pcm = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = AIU_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, + .period_bytes_min = AIU_FIFO_I2S_BLOCK, + .period_bytes_max = AIU_FIFO_I2S_BLOCK * USHRT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static int aiu_fifo_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + snd_soc_component_write(component, AIU_RST_SOFT, + AIU_RST_SOFT_I2S_FAST); + snd_soc_component_read(component, AIU_I2S_SYNC, &val); + break; + } + + return aiu_fifo_trigger(substream, cmd, dai); +} + +static int aiu_fifo_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_prepare(substream, dai); + if (ret) + return ret; + + snd_soc_component_update_bits(component, + AIU_MEM_I2S_BUF_CNTL, + AIU_MEM_I2S_BUF_CNTL_INIT, + AIU_MEM_I2S_BUF_CNTL_INIT); + snd_soc_component_update_bits(component, + AIU_MEM_I2S_BUF_CNTL, + AIU_MEM_I2S_BUF_CNTL_INIT, 0); + + return 0; +} + +static int aiu_fifo_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + unsigned int val; + int ret; + + ret = aiu_fifo_hw_params(substream, params, dai); + if (ret) + return ret; + + switch (params_physical_width(params)) { + case 16: + val = AIU_MEM_I2S_CONTROL_MODE_16BIT; + break; + case 32: + val = 0; + break; + default: + dev_err(dai->dev, "Unsupported physical width %u\n", + params_physical_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_MEM_I2S_CONTROL, + AIU_MEM_I2S_CONTROL_MODE_16BIT, + val); + + /* Setup the irq periodicity */ + val = params_period_bytes(params) / fifo->fifo_block; + val = FIELD_PREP(AIU_MEM_I2S_MASKS_IRQ_BLOCK, val); + snd_soc_component_update_bits(component, AIU_MEM_I2S_MASKS, + AIU_MEM_I2S_MASKS_IRQ_BLOCK, val); + + return 0; +} + +const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = { + .trigger = aiu_fifo_i2s_trigger, + .prepare = aiu_fifo_i2s_prepare, + .hw_params = aiu_fifo_i2s_hw_params, + .hw_free = aiu_fifo_hw_free, + .startup = aiu_fifo_startup, + .shutdown = aiu_fifo_shutdown, +}; + +int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + struct aiu_fifo *fifo; + int ret; + + ret = aiu_fifo_dai_probe(dai); + if (ret) + return ret; + + fifo = dai->playback_dma_data; + + fifo->pcm = &fifo_i2s_pcm; + fifo->mem_offset = AIU_MEM_I2S_START; + fifo->fifo_block = AIU_FIFO_I2S_BLOCK; + fifo->pclk = aiu->i2s.clks[PCLK].clk; + fifo->irq = aiu->i2s.irq; + + return 0; +} diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c new file mode 100644 index 000000000000..44eb6faacf44 --- /dev/null +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -0,0 +1,186 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_IEC958_DCU_FF_CTRL_EN BIT(0) +#define AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE BIT(1) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_MODE GENMASK(3, 2) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_OUT_THD BIT(2) +#define AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ BIT(3) +#define AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN BIT(4) +#define AIU_IEC958_DCU_FF_CTRL_BYTE_SEEK BIT(5) +#define AIU_IEC958_DCU_FF_CTRL_CONTINUE BIT(6) +#define AIU_MEM_IEC958_CONTROL_ENDIAN GENMASK(5, 3) +#define AIU_MEM_IEC958_CONTROL_RD_DDR BIT(6) +#define AIU_MEM_IEC958_CONTROL_MODE_16BIT BIT(7) +#define AIU_MEM_IEC958_CONTROL_MODE_LINEAR BIT(8) +#define AIU_MEM_IEC958_BUF_CNTL_INIT BIT(0) + +#define AIU_FIFO_SPDIF_BLOCK 8 + +static struct snd_pcm_hardware fifo_spdif_pcm = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = AIU_FORMATS, + .rate_min = 5512, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = AIU_FIFO_SPDIF_BLOCK, + .period_bytes_max = AIU_FIFO_SPDIF_BLOCK * USHRT_MAX, + .periods_min = 2, + .periods_max = UINT_MAX, + + /* No real justification for this */ + .buffer_bytes_max = 1 * 1024 * 1024, +}; + +static void fifo_spdif_dcu_enable(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL, + AIU_IEC958_DCU_FF_CTRL_EN, + enable ? AIU_IEC958_DCU_FF_CTRL_EN : 0); +} + +static int fifo_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_trigger(substream, cmd, dai); + if (ret) + return ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + fifo_spdif_dcu_enable(component, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + fifo_spdif_dcu_enable(component, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int fifo_spdif_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + int ret; + + ret = aiu_fifo_prepare(substream, dai); + if (ret) + return ret; + + snd_soc_component_update_bits(component, + AIU_MEM_IEC958_BUF_CNTL, + AIU_MEM_IEC958_BUF_CNTL_INIT, + AIU_MEM_IEC958_BUF_CNTL_INIT); + snd_soc_component_update_bits(component, + AIU_MEM_IEC958_BUF_CNTL, + AIU_MEM_IEC958_BUF_CNTL_INIT, 0); + + return 0; +} + +static int fifo_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + int ret; + + ret = aiu_fifo_hw_params(substream, params, dai); + if (ret) + return ret; + + val = AIU_MEM_IEC958_CONTROL_RD_DDR | + AIU_MEM_IEC958_CONTROL_MODE_LINEAR; + + switch (params_physical_width(params)) { + case 16: + val |= AIU_MEM_IEC958_CONTROL_MODE_16BIT; + break; + case 32: + break; + default: + dev_err(dai->dev, "Unsupported physical width %u\n", + params_physical_width(params)); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AIU_MEM_IEC958_CONTROL, + AIU_MEM_IEC958_CONTROL_ENDIAN | + AIU_MEM_IEC958_CONTROL_RD_DDR | + AIU_MEM_IEC958_CONTROL_MODE_LINEAR | + AIU_MEM_IEC958_CONTROL_MODE_16BIT, + val); + + /* Number bytes read by the FIFO between each IRQ */ + snd_soc_component_write(component, AIU_IEC958_BPF, + params_period_bytes(params)); + + /* + * AUTO_DISABLE and SYNC_HEAD are enabled by default but + * this should be disabled in PCM (uncompressed) mode + */ + snd_soc_component_update_bits(component, AIU_IEC958_DCU_FF_CTRL, + AIU_IEC958_DCU_FF_CTRL_AUTO_DISABLE | + AIU_IEC958_DCU_FF_CTRL_IRQ_MODE | + AIU_IEC958_DCU_FF_CTRL_SYNC_HEAD_EN, + AIU_IEC958_DCU_FF_CTRL_IRQ_FRAME_READ); + + return 0; +} + +const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = { + .trigger = fifo_spdif_trigger, + .prepare = fifo_spdif_prepare, + .hw_params = fifo_spdif_hw_params, + .hw_free = aiu_fifo_hw_free, + .startup = aiu_fifo_startup, + .shutdown = aiu_fifo_shutdown, +}; + +int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu *aiu = snd_soc_component_get_drvdata(component); + struct aiu_fifo *fifo; + int ret; + + ret = aiu_fifo_dai_probe(dai); + if (ret) + return ret; + + fifo = dai->playback_dma_data; + + fifo->pcm = &fifo_spdif_pcm; + fifo->mem_offset = AIU_MEM_IEC958_START; + fifo->fifo_block = 1; + fifo->pclk = aiu->spdif.clks[PCLK].clk; + fifo->irq = aiu->spdif.irq; + + return 0; +} diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c new file mode 100644 index 000000000000..d9cede4c33ff --- /dev/null +++ b/sound/soc/meson/aiu-fifo.c @@ -0,0 +1,223 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "aiu-fifo.h" + +#define AIU_MEM_START 0x00 +#define AIU_MEM_RD 0x04 +#define AIU_MEM_END 0x08 +#define AIU_MEM_MASKS 0x0c +#define AIU_MEM_MASK_CH_RD GENMASK(7, 0) +#define AIU_MEM_MASK_CH_MEM GENMASK(15, 8) +#define AIU_MEM_CONTROL 0x10 +#define AIU_MEM_CONTROL_INIT BIT(0) +#define AIU_MEM_CONTROL_FILL_EN BIT(1) +#define AIU_MEM_CONTROL_EMPTY_EN BIT(2) + +static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + + return asoc_rtd_to_cpu(rtd, 0); +} + +snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_dai *dai = aiu_fifo_dai(substream); + struct aiu_fifo *fifo = dai->playback_dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int addr; + + snd_soc_component_read(component, fifo->mem_offset + AIU_MEM_RD, + &addr); + + return bytes_to_frames(runtime, addr - (unsigned int)runtime->dma_addr); +} + +static void aiu_fifo_enable(struct snd_soc_dai *dai, bool enable) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + unsigned int en_mask = (AIU_MEM_CONTROL_FILL_EN | + AIU_MEM_CONTROL_EMPTY_EN); + + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + en_mask, enable ? en_mask : 0); +} + +int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + aiu_fifo_enable(dai, true); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_STOP: + aiu_fifo_enable(dai, false); + break; + default: + return -EINVAL; + } + + return 0; +} + +int aiu_fifo_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + AIU_MEM_CONTROL_INIT, + AIU_MEM_CONTROL_INIT); + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_CONTROL, + AIU_MEM_CONTROL_INIT, 0); + return 0; +} + +int aiu_fifo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_component *component = dai->component; + struct aiu_fifo *fifo = dai->playback_dma_data; + dma_addr_t end; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) + return ret; + + /* Setup the fifo boundaries */ + end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block; + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_START, + runtime->dma_addr); + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_RD, + runtime->dma_addr); + snd_soc_component_write(component, fifo->mem_offset + AIU_MEM_END, + end); + + /* Setup the fifo to read all the memory - no skip */ + snd_soc_component_update_bits(component, + fifo->mem_offset + AIU_MEM_MASKS, + AIU_MEM_MASK_CH_RD | AIU_MEM_MASK_CH_MEM, + FIELD_PREP(AIU_MEM_MASK_CH_RD, 0xff) | + FIELD_PREP(AIU_MEM_MASK_CH_MEM, 0xff)); + + return 0; +} + +int aiu_fifo_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static irqreturn_t aiu_fifo_isr(int irq, void *dev_id) +{ + struct snd_pcm_substream *playback = dev_id; + + snd_pcm_period_elapsed(playback); + + return IRQ_HANDLED; +} + +int aiu_fifo_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo = dai->playback_dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, fifo->pcm); + + /* + * Make sure the buffer and period size are multiple of the fifo burst + * size + */ + ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + fifo->fifo_block); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + fifo->fifo_block); + if (ret) + return ret; + + ret = clk_prepare_enable(fifo->pclk); + if (ret) + return ret; + + ret = request_irq(fifo->irq, aiu_fifo_isr, 0, dev_name(dai->dev), + substream); + if (ret) + clk_disable_unprepare(fifo->pclk); + + return ret; +} + +void aiu_fifo_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo = dai->playback_dma_data; + + free_irq(fifo->irq, substream); + clk_disable_unprepare(fifo->pclk); +} + +int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + struct snd_pcm_substream *substream = + rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_card *card = rtd->card->snd_card; + struct aiu_fifo *fifo = dai->playback_dma_data; + size_t size = fifo->pcm->buffer_bytes_max; + + snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); + + return 0; +} + +int aiu_fifo_dai_probe(struct snd_soc_dai *dai) +{ + struct aiu_fifo *fifo; + + fifo = kzalloc(sizeof(*fifo), GFP_KERNEL); + if (!fifo) + return -ENOMEM; + + dai->playback_dma_data = fifo; + + return 0; +} + +int aiu_fifo_dai_remove(struct snd_soc_dai *dai) +{ + kfree(dai->playback_dma_data); + + return 0; +} + diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h new file mode 100644 index 000000000000..42ce266677cc --- /dev/null +++ b/sound/soc/meson/aiu-fifo.h @@ -0,0 +1,50 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2020 BayLibre, SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AIU_FIFO_H +#define _MESON_AIU_FIFO_H + +struct snd_pcm_hardware; +struct snd_soc_component_driver; +struct snd_soc_dai_driver; +struct clk; +struct snd_pcm_ops; +struct snd_pcm_substream; +struct snd_soc_dai; +struct snd_pcm_hw_params; +struct platform_device; + +struct aiu_fifo { + struct snd_pcm_hardware *pcm; + unsigned int mem_offset; + unsigned int fifo_block; + struct clk *pclk; + int irq; +}; + +int aiu_fifo_dai_probe(struct snd_soc_dai *dai); +int aiu_fifo_dai_remove(struct snd_soc_dai *dai); + +snd_pcm_uframes_t aiu_fifo_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream); + +int aiu_fifo_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai); +int aiu_fifo_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +int aiu_fifo_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +void aiu_fifo_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); +int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); + +#endif /* _MESON_AIU_FIFO_H */ diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c new file mode 100644 index 000000000000..dc35ca79021c --- /dev/null +++ b/sound/soc/meson/aiu.c @@ -0,0 +1,388 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <linux/reset.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <dt-bindings/sound/meson-aiu.h> +#include "aiu.h" +#include "aiu-fifo.h" + +#define AIU_I2S_MISC_958_SRC_SHIFT 3 + +static const char * const aiu_spdif_encode_sel_texts[] = { + "SPDIF", "I2S", +}; + +static SOC_ENUM_SINGLE_DECL(aiu_spdif_encode_sel_enum, AIU_I2S_MISC, + AIU_I2S_MISC_958_SRC_SHIFT, + aiu_spdif_encode_sel_texts); + +static const struct snd_kcontrol_new aiu_spdif_encode_mux = + SOC_DAPM_ENUM("SPDIF Buffer Src", aiu_spdif_encode_sel_enum); + +static const struct snd_soc_dapm_widget aiu_cpu_dapm_widgets[] = { + SND_SOC_DAPM_MUX("SPDIF SRC SEL", SND_SOC_NOPM, 0, 0, + &aiu_spdif_encode_mux), +}; + +static const struct snd_soc_dapm_route aiu_cpu_dapm_routes[] = { + { "I2S Encoder Playback", NULL, "I2S FIFO Playback" }, + { "SPDIF SRC SEL", "SPDIF", "SPDIF FIFO Playback" }, + { "SPDIF SRC SEL", "I2S", "I2S FIFO Playback" }, + { "SPDIF Encoder Playback", NULL, "SPDIF SRC SEL" }, +}; + +int aiu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name, + unsigned int component_id) +{ + struct snd_soc_dai *dai; + int id; + + if (args->args_count != 2) + return -EINVAL; + + if (args->args[0] != component_id) + return -EINVAL; + + id = args->args[1]; + + if (id < 0 || id >= component->num_dai) + return -EINVAL; + + for_each_component_dais(component, dai) { + if (id == 0) + break; + id--; + } + + *dai_name = dai->driver->name; + + return 0; +} + +static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + return aiu_of_xlate_dai_name(component, args, dai_name, AIU_CPU); +} + +static int aiu_cpu_component_probe(struct snd_soc_component *component) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + + /* Required for the SPDIF Source control operation */ + return clk_prepare_enable(aiu->i2s.clks[PCLK].clk); +} + +static void aiu_cpu_component_remove(struct snd_soc_component *component) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + + clk_disable_unprepare(aiu->i2s.clks[PCLK].clk); +} + +static const struct snd_soc_component_driver aiu_cpu_component = { + .name = "AIU CPU", + .dapm_widgets = aiu_cpu_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aiu_cpu_dapm_widgets), + .dapm_routes = aiu_cpu_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aiu_cpu_dapm_routes), + .of_xlate_dai_name = aiu_cpu_of_xlate_dai_name, + .pointer = aiu_fifo_pointer, + .probe = aiu_cpu_component_probe, + .remove = aiu_cpu_component_remove, +}; + +static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = { + [CPU_I2S_FIFO] = { + .name = "I2S FIFO", + .playback = { + .stream_name = "I2S FIFO Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_fifo_i2s_dai_ops, + .pcm_new = aiu_fifo_pcm_new, + .probe = aiu_fifo_i2s_dai_probe, + .remove = aiu_fifo_dai_remove, + }, + [CPU_SPDIF_FIFO] = { + .name = "SPDIF FIFO", + .playback = { + .stream_name = "SPDIF FIFO Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_fifo_spdif_dai_ops, + .pcm_new = aiu_fifo_pcm_new, + .probe = aiu_fifo_spdif_dai_probe, + .remove = aiu_fifo_dai_remove, + }, + [CPU_I2S_ENCODER] = { + .name = "I2S Encoder", + .playback = { + .stream_name = "I2S Encoder Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = AIU_FORMATS, + }, + .ops = &aiu_encoder_i2s_dai_ops, + }, + [CPU_SPDIF_ENCODER] = { + .name = "SPDIF Encoder", + .playback = { + .stream_name = "SPDIF Encoder Playback", + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000), + .formats = AIU_FORMATS, + }, + .ops = &aiu_encoder_spdif_dai_ops, + } +}; + +static const struct regmap_config aiu_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0x2ac, +}; + +static int aiu_clk_bulk_get(struct device *dev, + const char * const *ids, + unsigned int num, + struct aiu_interface *interface) +{ + struct clk_bulk_data *clks; + int i, ret; + + clks = devm_kcalloc(dev, num, sizeof(*clks), GFP_KERNEL); + if (!clks) + return -ENOMEM; + + for (i = 0; i < num; i++) + clks[i].id = ids[i]; + + ret = devm_clk_bulk_get(dev, num, clks); + if (ret < 0) + return ret; + + interface->clks = clks; + interface->clk_num = num; + return 0; +} + +static const char * const aiu_i2s_ids[] = { + [PCLK] = "i2s_pclk", + [AOCLK] = "i2s_aoclk", + [MCLK] = "i2s_mclk", + [MIXER] = "i2s_mixer", +}; + +static const char * const aiu_spdif_ids[] = { + [PCLK] = "spdif_pclk", + [AOCLK] = "spdif_aoclk", + [MCLK] = "spdif_mclk_sel" +}; + +static int aiu_clk_get(struct device *dev) +{ + struct aiu *aiu = dev_get_drvdata(dev); + int ret; + + aiu->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(aiu->pclk)) { + if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) + dev_err(dev, "Can't get the aiu pclk\n"); + return PTR_ERR(aiu->pclk); + } + + aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); + if (IS_ERR(aiu->spdif_mclk)) { + if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) + dev_err(dev, "Can't get the aiu spdif master clock\n"); + return PTR_ERR(aiu->spdif_mclk); + } + + ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), + &aiu->i2s); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Can't get the i2s clocks\n"); + return ret; + } + + ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), + &aiu->spdif); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Can't get the spdif clocks\n"); + return ret; + } + + ret = clk_prepare_enable(aiu->pclk); + if (ret) { + dev_err(dev, "peripheral clock enable failed\n"); + return ret; + } + + ret = devm_add_action_or_reset(dev, + (void(*)(void *))clk_disable_unprepare, + aiu->pclk); + if (ret) + dev_err(dev, "failed to add reset action on pclk"); + + return ret; +} + +static int aiu_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + void __iomem *regs; + struct regmap *map; + struct aiu *aiu; + int ret; + + aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); + if (!aiu) + return -ENOMEM; + + aiu->platform = device_get_match_data(dev); + if (!aiu->platform) + return -ENODEV; + + platform_set_drvdata(pdev, aiu); + + ret = device_reset(dev); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to reset device\n"); + return ret; + } + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + map = devm_regmap_init_mmio(dev, regs, &aiu_regmap_cfg); + if (IS_ERR(map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(map)); + return PTR_ERR(map); + } + + aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s"); + if (aiu->i2s.irq < 0) + return aiu->i2s.irq; + + aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif"); + if (aiu->spdif.irq < 0) + return aiu->spdif.irq; + + ret = aiu_clk_get(dev); + if (ret) + return ret; + + /* Register the cpu component of the aiu */ + ret = snd_soc_register_component(dev, &aiu_cpu_component, + aiu_cpu_dai_drv, + ARRAY_SIZE(aiu_cpu_dai_drv)); + if (ret) { + dev_err(dev, "Failed to register cpu component\n"); + return ret; + } + + /* Register the hdmi codec control component */ + ret = aiu_hdmi_ctrl_register_component(dev); + if (ret) { + dev_err(dev, "Failed to register hdmi control component\n"); + goto err; + } + + /* Register the internal dac control component on gxl */ + if (aiu->platform->has_acodec) { + ret = aiu_acodec_ctrl_register_component(dev); + if (ret) { + dev_err(dev, + "Failed to register acodec control component\n"); + goto err; + } + } + + return 0; +err: + snd_soc_unregister_component(dev); + return ret; +} + +static int aiu_remove(struct platform_device *pdev) +{ + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + +static const struct aiu_platform_data aiu_gxbb_pdata = { + .has_acodec = false, + .has_clk_ctrl_more_i2s_div = true, +}; + +static const struct aiu_platform_data aiu_gxl_pdata = { + .has_acodec = true, + .has_clk_ctrl_more_i2s_div = true, +}; + +static const struct aiu_platform_data aiu_meson8_pdata = { + .has_acodec = false, + .has_clk_ctrl_more_i2s_div = false, +}; + +static const struct of_device_id aiu_of_match[] = { + { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, + { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, + {} +}; +MODULE_DEVICE_TABLE(of, aiu_of_match); + +static struct platform_driver aiu_pdrv = { + .probe = aiu_probe, + .remove = aiu_remove, + .driver = { + .name = "meson-aiu", + .of_match_table = aiu_of_match, + }, +}; +module_platform_driver(aiu_pdrv); + +MODULE_DESCRIPTION("Meson AIU Driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h new file mode 100644 index 000000000000..87aa19ac4af3 --- /dev/null +++ b/sound/soc/meson/aiu.h @@ -0,0 +1,89 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR MIT) */ +/* + * Copyright (c) 2018 BayLibre, SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_AIU_H +#define _MESON_AIU_H + +struct clk; +struct clk_bulk_data; +struct device; +struct of_phandle_args; +struct snd_soc_dai; +struct snd_soc_dai_ops; + +enum aiu_clk_ids { + PCLK = 0, + AOCLK, + MCLK, + MIXER +}; + +struct aiu_interface { + struct clk_bulk_data *clks; + unsigned int clk_num; + int irq; +}; + +struct aiu_platform_data { + bool has_acodec; + bool has_clk_ctrl_more_i2s_div; +}; + +struct aiu { + struct clk *pclk; + struct clk *spdif_mclk; + struct aiu_interface i2s; + struct aiu_interface spdif; + const struct aiu_platform_data *platform; +}; + +#define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +int aiu_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name, + unsigned int component_id); + +int aiu_hdmi_ctrl_register_component(struct device *dev); +int aiu_acodec_ctrl_register_component(struct device *dev); + +int aiu_fifo_i2s_dai_probe(struct snd_soc_dai *dai); +int aiu_fifo_spdif_dai_probe(struct snd_soc_dai *dai); + +extern const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops; +extern const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops; +extern const struct snd_soc_dai_ops aiu_encoder_i2s_dai_ops; +extern const struct snd_soc_dai_ops aiu_encoder_spdif_dai_ops; + +#define AIU_IEC958_BPF 0x000 +#define AIU_958_MISC 0x010 +#define AIU_IEC958_DCU_FF_CTRL 0x01c +#define AIU_958_CHSTAT_L0 0x020 +#define AIU_958_CHSTAT_L1 0x024 +#define AIU_958_CTRL 0x028 +#define AIU_I2S_SOURCE_DESC 0x034 +#define AIU_I2S_DAC_CFG 0x040 +#define AIU_I2S_SYNC 0x044 +#define AIU_I2S_MISC 0x048 +#define AIU_RST_SOFT 0x054 +#define AIU_CLK_CTRL 0x058 +#define AIU_CLK_CTRL_MORE 0x064 +#define AIU_CODEC_DAC_LRCLK_CTRL 0x0a0 +#define AIU_HDMI_CLK_DATA_CTRL 0x0a8 +#define AIU_ACODEC_CTRL 0x0b0 +#define AIU_958_CHSTAT_R0 0x0c0 +#define AIU_958_CHSTAT_R1 0x0c4 +#define AIU_MEM_I2S_START 0x180 +#define AIU_MEM_I2S_MASKS 0x18c +#define AIU_MEM_I2S_CONTROL 0x190 +#define AIU_MEM_IEC958_START 0x194 +#define AIU_MEM_IEC958_CONTROL 0x1a4 +#define AIU_MEM_I2S_BUF_CNTL 0x1d8 +#define AIU_MEM_IEC958_BUF_CNTL 0x1fc + +#endif /* _MESON_AIU_H */ diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 1f698adde506..af46845f4ef2 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -9,11 +9,7 @@ #include <sound/soc-dai.h> #include "axg-tdm.h" - -struct axg_card { - struct snd_soc_card card; - void **link_data; -}; +#include "meson-card.h" struct axg_dai_link_tdm_mask { u32 tx; @@ -41,161 +37,15 @@ static const struct snd_soc_pcm_stream codec_params = { .channels_max = 8, }; -#define PREFIX "amlogic," - -static int axg_card_reallocate_links(struct axg_card *priv, - unsigned int num_links) -{ - struct snd_soc_dai_link *links; - void **ldata; - - links = krealloc(priv->card.dai_link, - num_links * sizeof(*priv->card.dai_link), - GFP_KERNEL | __GFP_ZERO); - ldata = krealloc(priv->link_data, - num_links * sizeof(*priv->link_data), - GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } - - priv->card.dai_link = links; - priv->link_data = ldata; - priv->card.num_links = num_links; - return 0; -} - -static int axg_card_parse_dai(struct snd_soc_card *card, - struct device_node *node, - struct device_node **dai_of_node, - const char **dai_name) -{ - struct of_phandle_args args; - int ret; - - if (!dai_name || !dai_of_node || !node) - return -EINVAL; - - ret = of_parse_phandle_with_args(node, "sound-dai", - "#sound-dai-cells", 0, &args); - if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); - return ret; - } - *dai_of_node = args.np; - - return snd_soc_get_dai_name(&args, dai_name); -} - -static int axg_card_set_link_name(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node, - const char *prefix) -{ - char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", - prefix, node->full_name); - if (!name) - return -ENOMEM; - - link->name = name; - link->stream_name = name; - - return 0; -} - -static void axg_card_clean_references(struct axg_card *priv) -{ - struct snd_soc_card *card = &priv->card; - struct snd_soc_dai_link *link; - struct snd_soc_dai_link_component *codec; - struct snd_soc_aux_dev *aux; - int i, j; - - if (card->dai_link) { - for_each_card_prelinks(card, i, link) { - if (link->cpus) - of_node_put(link->cpus->of_node); - for_each_link_codecs(link, j, codec) - of_node_put(codec->of_node); - } - } - - if (card->aux_dev) { - for_each_card_pre_auxs(card, i, aux) - of_node_put(aux->dlc.of_node); - } - - kfree(card->dai_link); - kfree(priv->link_data); -} - -static int axg_card_add_aux_devices(struct snd_soc_card *card) -{ - struct device_node *node = card->dev->of_node; - struct snd_soc_aux_dev *aux; - int num, i; - - num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); - if (num == -ENOENT) { - /* - * It is ok to have no auxiliary devices but for this card it - * is a strange situtation. Let's warn the about it. - */ - dev_warn(card->dev, "card has no auxiliary devices\n"); - return 0; - } else if (num < 0) { - dev_err(card->dev, "error getting auxiliary devices: %d\n", - num); - return num; - } - - aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); - if (!aux) - return -ENOMEM; - card->aux_dev = aux; - card->num_aux_devs = num; - - for_each_card_pre_auxs(card, i, aux) { - aux->dlc.of_node = - of_parse_phandle(node, "audio-aux-devs", i); - if (!aux->dlc.of_node) - return -EINVAL; - } - - return 0; -} - static int axg_card_tdm_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; - struct snd_soc_dai *codec_dai; - unsigned int mclk; - int ret, i; - - if (be->mclk_fs) { - mclk = params_rate(params) * be->mclk_fs; - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - return ret; - } - - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - return ret; - } - return 0; + return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); } static const struct snd_soc_ops axg_card_tdm_be_ops = { @@ -204,13 +54,13 @@ static const struct snd_soc_ops axg_card_tdm_be_ops = { static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; struct snd_soc_dai *codec_dai; int ret, i; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, be->codec_masks[i].tx, be->codec_masks[i].rx, @@ -222,10 +72,10 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) } } - ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, be->tx_mask, be->rx_mask, + ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), be->tx_mask, be->rx_mask, be->slots, be->slot_width); if (ret) { - dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n"); return ret; } @@ -234,16 +84,16 @@ static int axg_card_tdm_dai_init(struct snd_soc_pcm_runtime *rtd) static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) { - struct axg_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); struct axg_dai_link_tdm_data *be = (struct axg_dai_link_tdm_data *)priv->link_data[rtd->num]; int ret; /* The loopback rx_mask is the pad tx_mask */ - ret = axg_tdm_set_tdm_slots(rtd->cpu_dai, NULL, be->tx_mask, + ret = axg_tdm_set_tdm_slots(asoc_rtd_to_cpu(rtd, 0), NULL, be->tx_mask, be->slots, be->slot_width); if (ret) { - dev_err(rtd->cpu_dai->dev, "setting tdm link slots failed\n"); + dev_err(asoc_rtd_to_cpu(rtd, 0)->dev, "setting tdm link slots failed\n"); return ret; } @@ -253,14 +103,14 @@ static int axg_card_tdm_dai_lb_init(struct snd_soc_pcm_runtime *rtd) static int axg_card_add_tdm_loopback(struct snd_soc_card *card, int *index) { - struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct meson_card *priv = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *pad = &card->dai_link[*index]; struct snd_soc_dai_link *lb; struct snd_soc_dai_link_component *dlc; int ret; /* extend links */ - ret = axg_card_reallocate_links(priv, card->num_links + 1); + ret = meson_card_reallocate_links(card, card->num_links + 1); if (ret) return ret; @@ -304,32 +154,6 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, return 0; } -static unsigned int axg_card_parse_daifmt(struct device_node *node, - struct device_node *cpu_node) -{ - struct device_node *bitclkmaster = NULL; - struct device_node *framemaster = NULL; - unsigned int daifmt; - - daifmt = snd_soc_of_parse_daifmt(node, PREFIX, - &bitclkmaster, &framemaster); - daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - - /* If no master is provided, default to cpu master */ - if (!bitclkmaster || bitclkmaster == cpu_node) { - daifmt |= (!framemaster || framemaster == cpu_node) ? - SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; - } else { - daifmt |= (!framemaster || framemaster == cpu_node) ? - SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; - } - - of_node_put(bitclkmaster); - of_node_put(framemaster); - - return daifmt; -} - static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, struct snd_soc_dai_link *link, struct device_node *node, @@ -424,7 +248,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, struct device_node *node, int *index) { - struct axg_card *priv = snd_soc_card_get_drvdata(card); + struct meson_card *priv = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *link = &card->dai_link[*index]; struct axg_dai_link_tdm_data *be; int ret; @@ -438,7 +262,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, /* Setup tdm link */ link->ops = &axg_card_tdm_be_ops; link->init = axg_card_tdm_dai_init; - link->dai_fmt = axg_card_parse_daifmt(node, link->cpus->of_node); + link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node); of_property_read_u32(node, "mclk-fs", &be->mclk_fs); @@ -462,97 +286,25 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, return 0; } -static int axg_card_set_be_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node) -{ - struct snd_soc_dai_link_component *codec; - struct device_node *np; - int ret, num_codecs; - - link->no_pcm = 1; - link->dpcm_playback = 1; - link->dpcm_capture = 1; - - num_codecs = of_get_child_count(node); - if (!num_codecs) { - dev_err(card->dev, "be link %s has no codec\n", - node->full_name); - return -EINVAL; - } - - codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - link->codecs = codec; - link->num_codecs = num_codecs; - - for_each_child_of_node(node, np) { - ret = axg_card_parse_dai(card, np, &codec->of_node, - &codec->dai_name); - if (ret) { - of_node_put(np); - return ret; - } - - codec++; - } - - ret = axg_card_set_link_name(card, link, node, "be"); - if (ret) - dev_err(card->dev, "error setting %pOFn link name\n", np); - - return ret; -} - -static int axg_card_set_fe_link(struct snd_soc_card *card, - struct snd_soc_dai_link *link, - struct device_node *node, - bool is_playback) -{ - struct snd_soc_dai_link_component *codec; - - codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL); - if (!codec) - return -ENOMEM; - - link->codecs = codec; - link->num_codecs = 1; - - link->dynamic = 1; - link->dpcm_merged_format = 1; - link->dpcm_merged_chan = 1; - link->dpcm_merged_rate = 1; - link->codecs->dai_name = "snd-soc-dummy-dai"; - link->codecs->name = "snd-soc-dummy"; - - if (is_playback) - link->dpcm_playback = 1; - else - link->dpcm_capture = 1; - - return axg_card_set_link_name(card, link, node, "fe"); -} - static int axg_card_cpu_is_capture_fe(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-toddr"); + return of_device_is_compatible(np, DT_PREFIX "axg-toddr"); } static int axg_card_cpu_is_playback_fe(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-frddr"); + return of_device_is_compatible(np, DT_PREFIX "axg-frddr"); } static int axg_card_cpu_is_tdm_iface(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "axg-tdm-iface"); + return of_device_is_compatible(np, DT_PREFIX "axg-tdm-iface"); } static int axg_card_cpu_is_codec(struct device_node *np) { - return of_device_is_compatible(np, PREFIX "g12a-tohdmitx"); + return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") || + of_device_is_compatible(np, DT_PREFIX "g12a-toacodec"); } static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, @@ -569,17 +321,17 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->cpus = cpu; dai_link->num_cpus = 1; - ret = axg_card_parse_dai(card, np, &dai_link->cpus->of_node, - &dai_link->cpus->dai_name); + ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node, + &dai_link->cpus->dai_name); if (ret) return ret; if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node)) - ret = axg_card_set_fe_link(card, dai_link, np, true); + ret = meson_card_set_fe_link(card, dai_link, np, true); else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node)) - ret = axg_card_set_fe_link(card, dai_link, np, false); + ret = meson_card_set_fe_link(card, dai_link, np, false); else - ret = axg_card_set_be_link(card, dai_link, np); + ret = meson_card_set_be_link(card, dai_link, np); if (ret) return ret; @@ -592,121 +344,21 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, return ret; } -static int axg_card_add_links(struct snd_soc_card *card) -{ - struct axg_card *priv = snd_soc_card_get_drvdata(card); - struct device_node *node = card->dev->of_node; - struct device_node *np; - int num, i, ret; - - num = of_get_child_count(node); - if (!num) { - dev_err(card->dev, "card has no links\n"); - return -EINVAL; - } - - ret = axg_card_reallocate_links(priv, num); - if (ret) - return ret; - - i = 0; - for_each_child_of_node(node, np) { - ret = axg_card_add_link(card, np, &i); - if (ret) { - of_node_put(np); - return ret; - } - - i++; - } - - return 0; -} - -static int axg_card_parse_of_optional(struct snd_soc_card *card, - const char *propname, - int (*func)(struct snd_soc_card *c, - const char *p)) -{ - /* If property is not provided, don't fail ... */ - if (!of_property_read_bool(card->dev->of_node, propname)) - return 0; - - /* ... but do fail if it is provided and the parsing fails */ - return func(card, propname); -} +static const struct meson_card_match_data axg_card_match_data = { + .add_link = axg_card_add_link, +}; static const struct of_device_id axg_card_of_match[] = { - { .compatible = "amlogic,axg-sound-card", }, - {} + { + .compatible = "amlogic,axg-sound-card", + .data = &axg_card_match_data, + }, {} }; MODULE_DEVICE_TABLE(of, axg_card_of_match); -static int axg_card_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct axg_card *priv; - int ret; - - priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - platform_set_drvdata(pdev, priv); - snd_soc_card_set_drvdata(&priv->card, priv); - - priv->card.owner = THIS_MODULE; - priv->card.dev = dev; - - ret = snd_soc_of_parse_card_name(&priv->card, "model"); - if (ret < 0) - return ret; - - ret = axg_card_parse_of_optional(&priv->card, "audio-routing", - snd_soc_of_parse_audio_routing); - if (ret) { - dev_err(dev, "error while parsing routing\n"); - return ret; - } - - ret = axg_card_parse_of_optional(&priv->card, "audio-widgets", - snd_soc_of_parse_audio_simple_widgets); - if (ret) { - dev_err(dev, "error while parsing widgets\n"); - return ret; - } - - ret = axg_card_add_links(&priv->card); - if (ret) - goto out_err; - - ret = axg_card_add_aux_devices(&priv->card); - if (ret) - goto out_err; - - ret = devm_snd_soc_register_card(dev, &priv->card); - if (ret) - goto out_err; - - return 0; - -out_err: - axg_card_clean_references(priv); - return ret; -} - -static int axg_card_remove(struct platform_device *pdev) -{ - struct axg_card *priv = platform_get_drvdata(pdev); - - axg_card_clean_references(priv); - - return 0; -} - static struct platform_driver axg_card_pdrv = { - .probe = axg_card_probe, - .remove = axg_card_remove, + .probe = meson_card_probe, + .remove = meson_card_remove, .driver = { .name = "axg-sound-card", .of_match_table = axg_card_of_match, diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index c12b0d5e8ebf..2e9b56b29d31 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -47,7 +47,7 @@ static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) { struct snd_soc_pcm_runtime *rtd = ss->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static struct axg_fifo *axg_fifo_data(struct snd_pcm_substream *ss) diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c new file mode 100644 index 000000000000..9339fabccb79 --- /dev/null +++ b/sound/soc/meson/g12a-toacodec.c @@ -0,0 +1,252 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/bitfield.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/reset.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include <dt-bindings/sound/meson-g12a-toacodec.h> +#include "axg-tdm.h" +#include "meson-codec-glue.h" + +#define G12A_TOACODEC_DRV_NAME "g12a-toacodec" + +#define TOACODEC_CTRL0 0x0 +#define CTRL0_ENABLE_SHIFT 31 +#define CTRL0_DAT_SEL_SHIFT 14 +#define CTRL0_DAT_SEL (0x3 << CTRL0_DAT_SEL_SHIFT) +#define CTRL0_LANE_SEL 12 +#define CTRL0_LRCLK_SEL GENMASK(9, 8) +#define CTRL0_BLK_CAP_INV BIT(7) +#define CTRL0_BCLK_O_INV BIT(6) +#define CTRL0_BCLK_SEL GENMASK(5, 4) +#define CTRL0_MCLK_SEL GENMASK(2, 0) + +#define TOACODEC_OUT_CHMAX 2 + +static const char * const g12a_toacodec_mux_texts[] = { + "I2S A", "I2S B", "I2S C", +}; + +static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_DAT_SEL, + FIELD_PREP(CTRL0_DAT_SEL, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL0_DAT_SEL | + CTRL0_LRCLK_SEL | + CTRL0_BCLK_SEL, + FIELD_PREP(CTRL0_DAT_SEL, mux) | + FIELD_PREP(CTRL0_LRCLK_SEL, mux) | + FIELD_PREP(CTRL0_BCLK_SEL, mux)); + + /* + * FIXME: + * On this soc, the glue gets the MCLK directly from the clock + * controller instead of going the through the TDM interface. + * + * Here we assume interface A uses clock A, etc ... While it is + * true for now, it could be different. Instead the glue should + * find out the clock used by the interface and select the same + * source. For that, we will need regmap backed clock mux which + * is a work in progress + */ + snd_soc_component_update_bits(component, e->reg, + CTRL0_MCLK_SEL, + FIELD_PREP(CTRL0_MCLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + +static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, + CTRL0_DAT_SEL_SHIFT, + g12a_toacodec_mux_texts); + +static const struct snd_kcontrol_new g12a_toacodec_mux = + SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum, + snd_soc_dapm_get_enum_double, + g12a_toacodec_mux_put_enum); + +static const struct snd_kcontrol_new g12a_toacodec_out_enable = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0, + CTRL0_ENABLE_SHIFT, 1, 0); + +static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = { + SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_mux), + SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_out_enable), +}; + +static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + int ret; + + ret = meson_codec_glue_input_hw_params(substream, params, dai); + if (ret) + return ret; + + /* The glue will provide 1 lane out of the 4 to the output */ + data = meson_codec_glue_input_get_data(dai); + data->params.channels_min = min_t(unsigned int, TOACODEC_OUT_CHMAX, + data->params.channels_min); + data->params.channels_max = min_t(unsigned int, TOACODEC_OUT_CHMAX, + data->params.channels_max); + + return 0; +} + +static const struct snd_soc_dai_ops g12a_toacodec_input_ops = { + .hw_params = g12a_toacodec_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, +}; + +static const struct snd_soc_dai_ops g12a_toacodec_output_ops = { + .startup = meson_codec_glue_output_startup, +}; + +#define TOACODEC_STREAM(xname, xsuffix, xchmax) \ +{ \ + .stream_name = xname " " xsuffix, \ + .channels_min = 1, \ + .channels_max = (xchmax), \ + .rate_min = 5512, \ + .rate_max = 192000, \ + .formats = AXG_TDM_FORMATS, \ +} + +#define TOACODEC_INPUT(xname, xid) { \ + .name = xname, \ + .id = (xid), \ + .playback = TOACODEC_STREAM(xname, "Playback", 8), \ + .ops = &g12a_toacodec_input_ops, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ +} + +#define TOACODEC_OUTPUT(xname, xid) { \ + .name = xname, \ + .id = (xid), \ + .capture = TOACODEC_STREAM(xname, "Capture", TOACODEC_OUT_CHMAX), \ + .ops = &g12a_toacodec_output_ops, \ +} + +static struct snd_soc_dai_driver g12a_toacodec_dai_drv[] = { + TOACODEC_INPUT("IN A", TOACODEC_IN_A), + TOACODEC_INPUT("IN B", TOACODEC_IN_B), + TOACODEC_INPUT("IN C", TOACODEC_IN_C), + TOACODEC_OUTPUT("OUT", TOACODEC_OUT), +}; + +static int g12a_toacodec_component_probe(struct snd_soc_component *c) +{ + /* Initialize the static clock parameters */ + return snd_soc_component_write(c, TOACODEC_CTRL0, + CTRL0_BLK_CAP_INV); +} + +static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { + { "SRC", "I2S A", "IN A Playback" }, + { "SRC", "I2S B", "IN B Playback" }, + { "SRC", "I2S C", "IN C Playback" }, + { "OUT EN", "Switch", "SRC" }, + { "OUT Capture", NULL, "OUT EN" }, +}; + +static const struct snd_kcontrol_new g12a_toacodec_controls[] = { + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), +}; + +static const struct snd_soc_component_driver g12a_toacodec_component_drv = { + .probe = g12a_toacodec_component_probe, + .controls = g12a_toacodec_controls, + .num_controls = ARRAY_SIZE(g12a_toacodec_controls), + .dapm_widgets = g12a_toacodec_widgets, + .num_dapm_widgets = ARRAY_SIZE(g12a_toacodec_widgets), + .dapm_routes = g12a_toacodec_routes, + .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config g12a_toacodec_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, +}; + +static const struct of_device_id g12a_toacodec_of_match[] = { + { .compatible = "amlogic,g12a-toacodec", }, + {} +}; +MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match); + +static int g12a_toacodec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + void __iomem *regs; + struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + map = devm_regmap_init_mmio(dev, regs, &g12a_toacodec_regmap_cfg); + if (IS_ERR(map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(map)); + return PTR_ERR(map); + } + + return devm_snd_soc_register_component(dev, + &g12a_toacodec_component_drv, g12a_toacodec_dai_drv, + ARRAY_SIZE(g12a_toacodec_dai_drv)); +} + +static struct platform_driver g12a_toacodec_pdrv = { + .driver = { + .name = G12A_TOACODEC_DRV_NAME, + .of_match_table = g12a_toacodec_of_match, + }, + .probe = g12a_toacodec_probe, +}; +module_platform_driver(g12a_toacodec_pdrv); + +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_DESCRIPTION("Amlogic G12a To Internal DAC Codec Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 8a0db28a6a40..9b2b59536ced 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -13,112 +13,51 @@ #include <sound/soc-dai.h> #include <dt-bindings/sound/meson-g12a-tohdmitx.h> +#include "meson-codec-glue.h" #define G12A_TOHDMITX_DRV_NAME "g12a-tohdmitx" #define TOHDMITX_CTRL0 0x0 #define CTRL0_ENABLE_SHIFT 31 -#define CTRL0_I2S_DAT_SEL GENMASK(13, 12) +#define CTRL0_I2S_DAT_SEL_SHIFT 12 +#define CTRL0_I2S_DAT_SEL (0x3 << CTRL0_I2S_DAT_SEL_SHIFT) #define CTRL0_I2S_LRCLK_SEL GENMASK(9, 8) #define CTRL0_I2S_BLK_CAP_INV BIT(7) #define CTRL0_I2S_BCLK_O_INV BIT(6) #define CTRL0_I2S_BCLK_SEL GENMASK(5, 4) #define CTRL0_SPDIF_CLK_CAP_INV BIT(3) #define CTRL0_SPDIF_CLK_O_INV BIT(2) -#define CTRL0_SPDIF_SEL BIT(1) +#define CTRL0_SPDIF_SEL_SHIFT 1 +#define CTRL0_SPDIF_SEL (0x1 << CTRL0_SPDIF_SEL_SHIFT) #define CTRL0_SPDIF_CLK_SEL BIT(0) -struct g12a_tohdmitx_input { - struct snd_soc_pcm_stream params; - unsigned int fmt; -}; - -static struct snd_soc_dapm_widget * -g12a_tohdmitx_get_input(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p = NULL; - struct snd_soc_dapm_widget *in; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (!p->connect) - continue; - - /* Check that we still are in the same component */ - if (snd_soc_dapm_to_component(w->dapm) != - snd_soc_dapm_to_component(p->source->dapm)) - continue; - - if (p->source->id == snd_soc_dapm_dai_in) - return p->source; - - in = g12a_tohdmitx_get_input(p->source); - if (in) - return in; - } - - return NULL; -} - -static struct g12a_tohdmitx_input * -g12a_tohdmitx_get_input_data(struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_widget *in = - g12a_tohdmitx_get_input(w); - struct snd_soc_dai *dai; - - if (WARN_ON(!in)) - return NULL; - - dai = in->priv; - - return dai->playback_dma_data; -} - static const char * const g12a_tohdmitx_i2s_mux_texts[] = { "I2S A", "I2S B", "I2S C", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_texts); - -static int g12a_tohdmitx_get_input_val(struct snd_soc_component *component, - unsigned int mask) -{ - unsigned int val; - - snd_soc_component_read(component, TOHDMITX_CTRL0, &val); - return (val & mask) >> __ffs(mask); -} - -static int g12a_tohdmitx_i2s_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_I2S_DAT_SEL); - - return 0; -} - static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_I2S_DAT_SEL); + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_I2S_DAT_SEL, + FIELD_PREP(CTRL0_I2S_DAT_SEL, + mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); - snd_soc_component_update_bits(component, TOHDMITX_CTRL0, + snd_soc_component_update_bits(component, e->reg, CTRL0_I2S_DAT_SEL | CTRL0_I2S_LRCLK_SEL | CTRL0_I2S_BCLK_SEL, @@ -131,30 +70,19 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0, + CTRL0_I2S_DAT_SEL_SHIFT, + g12a_tohdmitx_i2s_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = SOC_DAPM_ENUM_EXT("I2S Source", g12a_tohdmitx_i2s_mux_enum, - g12a_tohdmitx_i2s_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_i2s_mux_put_enum); static const char * const g12a_tohdmitx_spdif_mux_texts[] = { "SPDIF A", "SPDIF B", }; -static SOC_ENUM_SINGLE_EXT_DECL(g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_texts); - -static int g12a_tohdmitx_spdif_mux_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = - snd_soc_dapm_kcontrol_component(kcontrol); - - ucontrol->value.enumerated.item[0] = - g12a_tohdmitx_get_input_val(component, CTRL0_SPDIF_SEL); - - return 0; -} - static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -163,13 +91,18 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int mux = ucontrol->value.enumerated.item[0]; - unsigned int val = g12a_tohdmitx_get_input_val(component, - CTRL0_SPDIF_SEL); + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL, + FIELD_PREP(CTRL0_SPDIF_SEL, mux)); + + if (!changed) + return 0; /* Force disconnect of the mux while updating */ - if (val != mux) - snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); snd_soc_component_update_bits(component, TOHDMITX_CTRL0, CTRL0_SPDIF_SEL | @@ -182,9 +115,13 @@ static int g12a_tohdmitx_spdif_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_spdif_mux_enum, TOHDMITX_CTRL0, + CTRL0_SPDIF_SEL_SHIFT, + g12a_tohdmitx_spdif_mux_texts); + static const struct snd_kcontrol_new g12a_tohdmitx_spdif_mux = SOC_DAPM_ENUM_EXT("SPDIF Source", g12a_tohdmitx_spdif_mux_enum, - g12a_tohdmitx_spdif_mux_get_enum, + snd_soc_dapm_get_enum_double, g12a_tohdmitx_spdif_mux_put_enum); static const struct snd_kcontrol_new g12a_tohdmitx_out_enable = @@ -202,83 +139,13 @@ static const struct snd_soc_dapm_widget g12a_tohdmitx_widgets[] = { &g12a_tohdmitx_out_enable), }; -static int g12a_tohdmitx_input_probe(struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data; - - data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - dai->playback_dma_data = data; - return 0; -} - -static int g12a_tohdmitx_input_remove(struct snd_soc_dai *dai) -{ - kfree(dai->playback_dma_data); - return 0; -} - -static int g12a_tohdmitx_input_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); - data->params.rate_min = params_rate(params); - data->params.rate_max = params_rate(params); - data->params.formats = 1 << params_format(params); - data->params.channels_min = params_channels(params); - data->params.channels_max = params_channels(params); - data->params.sig_bits = dai->driver->playback.sig_bits; - - return 0; -} - - -static int g12a_tohdmitx_input_set_fmt(struct snd_soc_dai *dai, - unsigned int fmt) -{ - struct g12a_tohdmitx_input *data = dai->playback_dma_data; - - /* Save the source stream format for the downstream link */ - data->fmt = fmt; - return 0; -} - -static int g12a_tohdmitx_output_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct g12a_tohdmitx_input *in_data = - g12a_tohdmitx_get_input_data(dai->capture_widget); - - if (!in_data) - return -ENODEV; - - if (WARN_ON(!rtd->dai_link->params)) { - dev_warn(dai->dev, "codec2codec link expected\n"); - return -EINVAL; - } - - /* Replace link params with the input params */ - rtd->dai_link->params = &in_data->params; - - if (!in_data->fmt) - return 0; - - return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); -} - static const struct snd_soc_dai_ops g12a_tohdmitx_input_ops = { - .hw_params = g12a_tohdmitx_input_hw_params, - .set_fmt = g12a_tohdmitx_input_set_fmt, + .hw_params = meson_codec_glue_input_hw_params, + .set_fmt = meson_codec_glue_input_set_fmt, }; static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { - .startup = g12a_tohdmitx_output_startup, + .startup = meson_codec_glue_output_startup, }; #define TOHDMITX_SPDIF_FORMATS \ @@ -305,8 +172,8 @@ static const struct snd_soc_dai_ops g12a_tohdmitx_output_ops = { .id = (xid), \ .playback = TOHDMITX_STREAM(xname, "Playback", xfmt, xchmax), \ .ops = &g12a_tohdmitx_input_ops, \ - .probe = g12a_tohdmitx_input_probe, \ - .remove = g12a_tohdmitx_input_remove, \ + .probe = meson_codec_glue_input_dai_probe, \ + .remove = meson_codec_glue_input_dai_remove, \ } #define TOHDMITX_OUT(xname, xid, xfmt, xchmax) { \ diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c new file mode 100644 index 000000000000..7b01dcb73e5e --- /dev/null +++ b/sound/soc/meson/gx-card.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "meson-card.h" + +struct gx_dai_link_i2s_data { + unsigned int mclk_fs; +}; + +/* + * Base params for the codec to codec links + * Those will be over-written by the CPU side of the link + */ +static const struct snd_soc_pcm_stream codec_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 5525, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 8, +}; + +static int gx_card_i2s_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct meson_card *priv = snd_soc_card_get_drvdata(rtd->card); + struct gx_dai_link_i2s_data *be = + (struct gx_dai_link_i2s_data *)priv->link_data[rtd->num]; + + return meson_card_i2s_set_sysclk(substream, params, be->mclk_fs); +} + +static const struct snd_soc_ops gx_card_i2s_be_ops = { + .hw_params = gx_card_i2s_be_hw_params, +}; + +static int gx_card_parse_i2s(struct snd_soc_card *card, + struct device_node *node, + int *index) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = &card->dai_link[*index]; + struct gx_dai_link_i2s_data *be; + + /* Allocate i2s link parameters */ + be = devm_kzalloc(card->dev, sizeof(*be), GFP_KERNEL); + if (!be) + return -ENOMEM; + priv->link_data[*index] = be; + + /* Setup i2s link */ + link->ops = &gx_card_i2s_be_ops; + link->dai_fmt = meson_card_parse_daifmt(node, link->cpus->of_node); + + of_property_read_u32(node, "mclk-fs", &be->mclk_fs); + + return 0; +} + +static int gx_card_cpu_identify(struct snd_soc_dai_link_component *c, + char *match) +{ + if (of_device_is_compatible(c->of_node, DT_PREFIX "aiu")) { + if (strstr(c->dai_name, match)) + return 1; + } + + /* dai not matched */ + return 0; +} + +static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, + int *index) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[*index]; + struct snd_soc_dai_link_component *cpu; + int ret; + + cpu = devm_kzalloc(card->dev, sizeof(*cpu), GFP_KERNEL); + if (!cpu) + return -ENOMEM; + + dai_link->cpus = cpu; + dai_link->num_cpus = 1; + + ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node, + &dai_link->cpus->dai_name); + if (ret) + return ret; + + if (gx_card_cpu_identify(dai_link->cpus, "FIFO")) + ret = meson_card_set_fe_link(card, dai_link, np, true); + else + ret = meson_card_set_be_link(card, dai_link, np); + + if (ret) + return ret; + + /* Check if the cpu is the i2s encoder and parse i2s data */ + if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) + ret = gx_card_parse_i2s(card, np, index); + + /* Or apply codec to codec params if necessary */ + else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) + dai_link->params = &codec_params; + + return ret; +} + +static const struct meson_card_match_data gx_card_match_data = { + .add_link = gx_card_add_link, +}; + +static const struct of_device_id gx_card_of_match[] = { + { + .compatible = "amlogic,gx-sound-card", + .data = &gx_card_match_data, + }, {} +}; +MODULE_DEVICE_TABLE(of, gx_card_of_match); + +static struct platform_driver gx_card_pdrv = { + .probe = meson_card_probe, + .remove = meson_card_remove, + .driver = { + .name = "gx-sound-card", + .of_match_table = gx_card_of_match, + }, +}; +module_platform_driver(gx_card_pdrv); + +MODULE_DESCRIPTION("Amlogic GX ALSA machine driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c new file mode 100644 index 000000000000..2ca8c98e204f --- /dev/null +++ b/sound/soc/meson/meson-card-utils.c @@ -0,0 +1,385 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <sound/soc.h> + +#include "meson-card.h" + +int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + unsigned int mclk_fs) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + unsigned int mclk; + int ret, i; + + if (!mclk_fs) + return 0; + + mclk = params_rate(params) * mclk_fs; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + return ret; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_i2s_set_sysclk); + +int meson_card_reallocate_links(struct snd_soc_card *card, + unsigned int num_links) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *links; + void **ldata; + + links = krealloc(priv->card.dai_link, + num_links * sizeof(*priv->card.dai_link), + GFP_KERNEL | __GFP_ZERO); + ldata = krealloc(priv->link_data, + num_links * sizeof(*priv->link_data), + GFP_KERNEL | __GFP_ZERO); + + if (!links || !ldata) { + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } + + priv->card.dai_link = links; + priv->link_data = ldata; + priv->card.num_links = num_links; + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_reallocate_links); + +int meson_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + if (!dai_name || !dai_of_node || !node) + return -EINVAL; + + ret = of_parse_phandle_with_args(node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "can't parse dai %d\n", ret); + return ret; + } + *dai_of_node = args.np; + + return snd_soc_get_dai_name(&args, dai_name); +} +EXPORT_SYMBOL_GPL(meson_card_parse_dai); + +static int meson_card_set_link_name(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + const char *prefix) +{ + char *name = devm_kasprintf(card->dev, GFP_KERNEL, "%s.%s", + prefix, node->full_name); + if (!name) + return -ENOMEM; + + link->name = name; + link->stream_name = name; + + return 0; +} + +unsigned int meson_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node) +{ + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + /* If no master is provided, default to cpu master */ + if (!bitclkmaster || bitclkmaster == cpu_node) { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBS_CFM; + } else { + daifmt |= (!framemaster || framemaster == cpu_node) ? + SND_SOC_DAIFMT_CBM_CFS : SND_SOC_DAIFMT_CBM_CFM; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return daifmt; +} +EXPORT_SYMBOL_GPL(meson_card_parse_daifmt); + +int meson_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node) +{ + struct snd_soc_dai_link_component *codec; + struct device_node *np; + int ret, num_codecs; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + num_codecs = of_get_child_count(node); + if (!num_codecs) { + dev_err(card->dev, "be link %s has no codec\n", + node->full_name); + return -EINVAL; + } + + codec = devm_kcalloc(card->dev, num_codecs, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = num_codecs; + + for_each_child_of_node(node, np) { + ret = meson_card_parse_dai(card, np, &codec->of_node, + &codec->dai_name); + if (ret) { + of_node_put(np); + return ret; + } + + codec++; + } + + ret = meson_card_set_link_name(card, link, node, "be"); + if (ret) + dev_err(card->dev, "error setting %pOFn link name\n", np); + + return ret; +} +EXPORT_SYMBOL_GPL(meson_card_set_be_link); + +int meson_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + bool is_playback) +{ + struct snd_soc_dai_link_component *codec; + + codec = devm_kzalloc(card->dev, sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + + link->codecs = codec; + link->num_codecs = 1; + + link->dynamic = 1; + link->dpcm_merged_format = 1; + link->dpcm_merged_chan = 1; + link->dpcm_merged_rate = 1; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; + + if (is_playback) + link->dpcm_playback = 1; + else + link->dpcm_capture = 1; + + return meson_card_set_link_name(card, link, node, "fe"); +} +EXPORT_SYMBOL_GPL(meson_card_set_fe_link); + +static int meson_card_add_links(struct snd_soc_card *card) +{ + struct meson_card *priv = snd_soc_card_get_drvdata(card); + struct device_node *node = card->dev->of_node; + struct device_node *np; + int num, i, ret; + + num = of_get_child_count(node); + if (!num) { + dev_err(card->dev, "card has no links\n"); + return -EINVAL; + } + + ret = meson_card_reallocate_links(card, num); + if (ret) + return ret; + + i = 0; + for_each_child_of_node(node, np) { + ret = priv->match_data->add_link(card, np, &i); + if (ret) { + of_node_put(np); + return ret; + } + + i++; + } + + return 0; +} + +static int meson_card_parse_of_optional(struct snd_soc_card *card, + const char *propname, + int (*func)(struct snd_soc_card *c, + const char *p)) +{ + /* If property is not provided, don't fail ... */ + if (!of_property_read_bool(card->dev->of_node, propname)) + return 0; + + /* ... but do fail if it is provided and the parsing fails */ + return func(card, propname); +} + +static int meson_card_add_aux_devices(struct snd_soc_card *card) +{ + struct device_node *node = card->dev->of_node; + struct snd_soc_aux_dev *aux; + int num, i; + + num = of_count_phandle_with_args(node, "audio-aux-devs", NULL); + if (num == -ENOENT) { + return 0; + } else if (num < 0) { + dev_err(card->dev, "error getting auxiliary devices: %d\n", + num); + return num; + } + + aux = devm_kcalloc(card->dev, num, sizeof(*aux), GFP_KERNEL); + if (!aux) + return -ENOMEM; + card->aux_dev = aux; + card->num_aux_devs = num; + + for_each_card_pre_auxs(card, i, aux) { + aux->dlc.of_node = + of_parse_phandle(node, "audio-aux-devs", i); + if (!aux->dlc.of_node) + return -EINVAL; + } + + return 0; +} + +static void meson_card_clean_references(struct meson_card *priv) +{ + struct snd_soc_card *card = &priv->card; + struct snd_soc_dai_link *link; + struct snd_soc_dai_link_component *codec; + struct snd_soc_aux_dev *aux; + int i, j; + + if (card->dai_link) { + for_each_card_prelinks(card, i, link) { + if (link->cpus) + of_node_put(link->cpus->of_node); + for_each_link_codecs(link, j, codec) + of_node_put(codec->of_node); + } + } + + if (card->aux_dev) { + for_each_card_pre_auxs(card, i, aux) + of_node_put(aux->dlc.of_node); + } + + kfree(card->dai_link); + kfree(priv->link_data); +} + +int meson_card_probe(struct platform_device *pdev) +{ + const struct meson_card_match_data *data; + struct device *dev = &pdev->dev; + struct meson_card *priv; + int ret; + + data = of_device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + platform_set_drvdata(pdev, priv); + snd_soc_card_set_drvdata(&priv->card, priv); + + priv->card.owner = THIS_MODULE; + priv->card.dev = dev; + priv->match_data = data; + + ret = snd_soc_of_parse_card_name(&priv->card, "model"); + if (ret < 0) + return ret; + + ret = meson_card_parse_of_optional(&priv->card, "audio-routing", + snd_soc_of_parse_audio_routing); + if (ret) { + dev_err(dev, "error while parsing routing\n"); + return ret; + } + + ret = meson_card_parse_of_optional(&priv->card, "audio-widgets", + snd_soc_of_parse_audio_simple_widgets); + if (ret) { + dev_err(dev, "error while parsing widgets\n"); + return ret; + } + + ret = meson_card_add_links(&priv->card); + if (ret) + goto out_err; + + ret = meson_card_add_aux_devices(&priv->card); + if (ret) + goto out_err; + + ret = devm_snd_soc_register_card(dev, &priv->card); + if (ret) + goto out_err; + + return 0; + +out_err: + meson_card_clean_references(priv); + return ret; +} +EXPORT_SYMBOL_GPL(meson_card_probe); + +int meson_card_remove(struct platform_device *pdev) +{ + struct meson_card *priv = platform_get_drvdata(pdev); + + meson_card_clean_references(priv); + + return 0; +} +EXPORT_SYMBOL_GPL(meson_card_remove); + +MODULE_DESCRIPTION("Amlogic Sound Card Utils"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/meson/meson-card.h b/sound/soc/meson/meson-card.h new file mode 100644 index 000000000000..74314071c80d --- /dev/null +++ b/sound/soc/meson/meson-card.h @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2020 BayLibre, SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_SND_CARD_H +#define _MESON_SND_CARD_H + +struct device_node; +struct platform_device; + +struct snd_soc_card; +struct snd_pcm_substream; +struct snd_pcm_hw_params; + +#define DT_PREFIX "amlogic," + +struct meson_card_match_data { + int (*add_link)(struct snd_soc_card *card, + struct device_node *node, + int *index); +}; + +struct meson_card { + const struct meson_card_match_data *match_data; + struct snd_soc_card card; + void **link_data; +}; + +unsigned int meson_card_parse_daifmt(struct device_node *node, + struct device_node *cpu_node); + +int meson_card_i2s_set_sysclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + unsigned int mclk_fs); + +int meson_card_reallocate_links(struct snd_soc_card *card, + unsigned int num_links); +int meson_card_parse_dai(struct snd_soc_card *card, + struct device_node *node, + struct device_node **dai_of_node, + const char **dai_name); +int meson_card_set_be_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node); +int meson_card_set_fe_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link, + struct device_node *node, + bool is_playback); + +int meson_card_probe(struct platform_device *pdev); +int meson_card_remove(struct platform_device *pdev); + +#endif /* _MESON_SND_CARD_H */ diff --git a/sound/soc/meson/meson-codec-glue.c b/sound/soc/meson/meson-codec-glue.c new file mode 100644 index 000000000000..524a33472337 --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.c @@ -0,0 +1,149 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2019 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "meson-codec-glue.h" + +static struct snd_soc_dapm_widget * +meson_codec_glue_get_input(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p = NULL; + struct snd_soc_dapm_widget *in; + + snd_soc_dapm_widget_for_each_source_path(w, p) { + if (!p->connect) + continue; + + /* Check that we still are in the same component */ + if (snd_soc_dapm_to_component(w->dapm) != + snd_soc_dapm_to_component(p->source->dapm)) + continue; + + if (p->source->id == snd_soc_dapm_dai_in) + return p->source; + + in = meson_codec_glue_get_input(p->source); + if (in) + return in; + } + + return NULL; +} + +static void meson_codec_glue_input_set_data(struct snd_soc_dai *dai, + struct meson_codec_glue_input *data) +{ + dai->playback_dma_data = data; +} + +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai) +{ + return dai->playback_dma_data; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_get_data); + +static struct meson_codec_glue_input * +meson_codec_glue_output_get_input_data(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *in = + meson_codec_glue_get_input(w); + struct snd_soc_dai *dai; + + if (WARN_ON(!in)) + return NULL; + + dai = in->priv; + + return meson_codec_glue_input_get_data(dai); +} + +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + data->params.rates = snd_pcm_rate_to_rate_bit(params_rate(params)); + data->params.rate_min = params_rate(params); + data->params.rate_max = params_rate(params); + data->params.formats = 1ULL << (__force int) params_format(params); + data->params.channels_min = params_channels(params); + data->params.channels_max = params_channels(params); + data->params.sig_bits = dai->driver->playback.sig_bits; + + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_hw_params); + +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + /* Save the source stream format for the downstream link */ + data->fmt = fmt; + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_set_fmt); + +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct meson_codec_glue_input *in_data = + meson_codec_glue_output_get_input_data(dai->capture_widget); + + if (!in_data) + return -ENODEV; + + if (WARN_ON(!rtd->dai_link->params)) { + dev_warn(dai->dev, "codec2codec link expected\n"); + return -EINVAL; + } + + /* Replace link params with the input params */ + rtd->dai_link->params = &in_data->params; + + if (!in_data->fmt) + return 0; + + return snd_soc_runtime_set_dai_fmt(rtd, in_data->fmt); +} +EXPORT_SYMBOL_GPL(meson_codec_glue_output_startup); + +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data; + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + meson_codec_glue_input_set_data(dai, data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_probe); + +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai) +{ + struct meson_codec_glue_input *data = + meson_codec_glue_input_get_data(dai); + + kfree(data); + return 0; +} +EXPORT_SYMBOL_GPL(meson_codec_glue_input_dai_remove); + +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_DESCRIPTION("Amlogic Codec Glue Helpers"); +MODULE_LICENSE("GPL v2"); + diff --git a/sound/soc/meson/meson-codec-glue.h b/sound/soc/meson/meson-codec-glue.h new file mode 100644 index 000000000000..07f99446c0c6 --- /dev/null +++ b/sound/soc/meson/meson-codec-glue.h @@ -0,0 +1,32 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Copyright (c) 2018 Baylibre SAS. + * Author: Jerome Brunet <jbrunet@baylibre.com> + */ + +#ifndef _MESON_CODEC_GLUE_H +#define _MESON_CODEC_GLUE_H + +#include <sound/soc.h> + +struct meson_codec_glue_input { + struct snd_soc_pcm_stream params; + unsigned int fmt; +}; + +/* Input helpers */ +struct meson_codec_glue_input * +meson_codec_glue_input_get_data(struct snd_soc_dai *dai); +int meson_codec_glue_input_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +int meson_codec_glue_input_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt); +int meson_codec_glue_input_dai_probe(struct snd_soc_dai *dai); +int meson_codec_glue_input_dai_remove(struct snd_soc_dai *dai); + +/* Output helpers */ +int meson_codec_glue_output_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai); + +#endif /* _MESON_CODEC_GLUE_H */ diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c new file mode 100644 index 000000000000..56d2592c16d5 --- /dev/null +++ b/sound/soc/meson/t9015.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright (c) 2020 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet@baylibre.com> + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/reset.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define BLOCK_EN 0x00 +#define LORN_EN 0 +#define LORP_EN 1 +#define LOLN_EN 2 +#define LOLP_EN 3 +#define DACR_EN 4 +#define DACL_EN 5 +#define DACR_INV 20 +#define DACL_INV 21 +#define DACR_SRC 22 +#define DACL_SRC 23 +#define REFP_BUF_EN BIT(12) +#define BIAS_CURRENT_EN BIT(13) +#define VMID_GEN_FAST BIT(14) +#define VMID_GEN_EN BIT(15) +#define I2S_MODE BIT(30) +#define VOL_CTRL0 0x04 +#define GAIN_H 31 +#define GAIN_L 23 +#define VOL_CTRL1 0x08 +#define DAC_MONO 8 +#define RAMP_RATE 10 +#define VC_RAMP_MODE 12 +#define MUTE_MODE 13 +#define UNMUTE_MODE 14 +#define DAC_SOFT_MUTE 15 +#define DACR_VC 16 +#define DACL_VC 24 +#define LINEOUT_CFG 0x0c +#define LORN_POL 0 +#define LORP_POL 4 +#define LOLN_POL 8 +#define LOLP_POL 12 +#define POWER_CFG 0x10 + +struct t9015 { + struct clk *pclk; + struct regulator *avdd; +}; + +static int t9015_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int val; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + val = I2S_MODE; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + val = 0; + break; + + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, BLOCK_EN, I2S_MODE, val); + + if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) && + ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_LEFT_J)) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops t9015_dai_ops = { + .set_fmt = t9015_dai_set_fmt, +}; + +static struct snd_soc_dai_driver t9015_dai = { + .name = "t9015-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = (SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_LE | + SNDRV_PCM_FMTBIT_S24_LE), + }, + .ops = &t9015_dai_ops, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -9525, 0); + +static const char * const ramp_rate_txt[] = { "Fast", "Slow" }; +static SOC_ENUM_SINGLE_DECL(ramp_rate_enum, VOL_CTRL1, RAMP_RATE, + ramp_rate_txt); + +static const char * const dacr_in_txt[] = { "Right", "Left" }; +static SOC_ENUM_SINGLE_DECL(dacr_in_enum, BLOCK_EN, DACR_SRC, dacr_in_txt); + +static const char * const dacl_in_txt[] = { "Left", "Right" }; +static SOC_ENUM_SINGLE_DECL(dacl_in_enum, BLOCK_EN, DACL_SRC, dacl_in_txt); + +static const char * const mono_txt[] = { "Stereo", "Mono"}; +static SOC_ENUM_SINGLE_DECL(mono_enum, VOL_CTRL1, DAC_MONO, mono_txt); + +static const struct snd_kcontrol_new t9015_snd_controls[] = { + /* Volume Controls */ + SOC_ENUM("Playback Channel Mode", mono_enum), + SOC_SINGLE("Playback Switch", VOL_CTRL1, DAC_SOFT_MUTE, 1, 1), + SOC_DOUBLE_TLV("Playback Volume", VOL_CTRL1, DACL_VC, DACR_VC, + 0xff, 0, dac_vol_tlv), + + /* Ramp Controls */ + SOC_ENUM("Ramp Rate", ramp_rate_enum), + SOC_SINGLE("Volume Ramp Switch", VOL_CTRL1, VC_RAMP_MODE, 1, 0), + SOC_SINGLE("Mute Ramp Switch", VOL_CTRL1, MUTE_MODE, 1, 0), + SOC_SINGLE("Unmute Ramp Switch", VOL_CTRL1, UNMUTE_MODE, 1, 0), +}; + +static const struct snd_kcontrol_new t9015_right_dac_mux = + SOC_DAPM_ENUM("Right DAC Source", dacr_in_enum); +static const struct snd_kcontrol_new t9015_left_dac_mux = + SOC_DAPM_ENUM("Left DAC Source", dacl_in_enum); + +static const struct snd_soc_dapm_widget t9015_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("Right IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("Left IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("Right DAC Sel", SND_SOC_NOPM, 0, 0, + &t9015_right_dac_mux), + SND_SOC_DAPM_MUX("Left DAC Sel", SND_SOC_NOPM, 0, 0, + &t9015_left_dac_mux), + SND_SOC_DAPM_DAC("Right DAC", NULL, BLOCK_EN, DACR_EN, 0), + SND_SOC_DAPM_DAC("Left DAC", NULL, BLOCK_EN, DACL_EN, 0), + SND_SOC_DAPM_OUT_DRV("Right- Driver", BLOCK_EN, LORN_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Right+ Driver", BLOCK_EN, LORP_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Left- Driver", BLOCK_EN, LOLN_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("Left+ Driver", BLOCK_EN, LOLP_EN, 0, + NULL, 0), + SND_SOC_DAPM_OUTPUT("LORN"), + SND_SOC_DAPM_OUTPUT("LORP"), + SND_SOC_DAPM_OUTPUT("LOLN"), + SND_SOC_DAPM_OUTPUT("LOLP"), +}; + +static const struct snd_soc_dapm_route t9015_dapm_routes[] = { + { "Right IN", NULL, "Playback" }, + { "Left IN", NULL, "Playback" }, + { "Right DAC Sel", "Right", "Right IN" }, + { "Right DAC Sel", "Left", "Left IN" }, + { "Left DAC Sel", "Right", "Right IN" }, + { "Left DAC Sel", "Left", "Left IN" }, + { "Right DAC", NULL, "Right DAC Sel" }, + { "Left DAC", NULL, "Left DAC Sel" }, + { "Right- Driver", NULL, "Right DAC" }, + { "Right+ Driver", NULL, "Right DAC" }, + { "Left- Driver", NULL, "Left DAC" }, + { "Left+ Driver", NULL, "Left DAC" }, + { "LORN", NULL, "Right- Driver", }, + { "LORP", NULL, "Right+ Driver", }, + { "LOLN", NULL, "Left- Driver", }, + { "LOLP", NULL, "Left+ Driver", }, +}; + +static int t9015_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct t9015 *priv = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level now = + snd_soc_component_get_bias_level(component); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, BLOCK_EN, + BIAS_CURRENT_EN, + BIAS_CURRENT_EN); + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, BLOCK_EN, + BIAS_CURRENT_EN, + 0); + break; + case SND_SOC_BIAS_STANDBY: + ret = regulator_enable(priv->avdd); + if (ret) { + dev_err(component->dev, "AVDD enable failed\n"); + return ret; + } + + if (now == SND_SOC_BIAS_OFF) { + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN); + + mdelay(200); + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_FAST, + 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, BLOCK_EN, + VMID_GEN_EN | VMID_GEN_FAST | REFP_BUF_EN, + 0); + + regulator_disable(priv->avdd); + break; + } + + return 0; +} + +static const struct snd_soc_component_driver t9015_codec_driver = { + .set_bias_level = t9015_set_bias_level, + .controls = t9015_snd_controls, + .num_controls = ARRAY_SIZE(t9015_snd_controls), + .dapm_widgets = t9015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(t9015_dapm_widgets), + .dapm_routes = t9015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(t9015_dapm_routes), + .suspend_bias_off = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config t9015_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = POWER_CFG, +}; + +static int t9015_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct t9015 *priv; + void __iomem *regs; + struct regmap *regmap; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) + dev_err(dev, "failed to get core clock\n"); + return PTR_ERR(priv->pclk); + } + + priv->avdd = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(priv->avdd)) { + if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) + dev_err(dev, "failed to AVDD\n"); + return PTR_ERR(priv->avdd); + } + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dev, "core clock enable failed\n"); + return ret; + } + + ret = devm_add_action_or_reset(dev, + (void(*)(void *))clk_disable_unprepare, + priv->pclk); + if (ret) + return ret; + + ret = device_reset(dev); + if (ret) { + dev_err(dev, "reset failed\n"); + return ret; + } + + regs = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(regs)) { + dev_err(dev, "register map failed\n"); + return PTR_ERR(regs); + } + + regmap = devm_regmap_init_mmio(dev, regs, &t9015_regmap_config); + if (IS_ERR(regmap)) { + dev_err(dev, "regmap init failed\n"); + return PTR_ERR(regmap); + } + + /* + * Initialize output polarity: + * ATM the output polarity is fixed but in the future it might useful + * to add DT property to set this depending on the platform needs + */ + regmap_write(regmap, LINEOUT_CFG, 0x1111); + + return devm_snd_soc_register_component(dev, &t9015_codec_driver, + &t9015_dai, 1); +} + +static const struct of_device_id t9015_ids[] = { + { .compatible = "amlogic,t9015", }, + { } +}; +MODULE_DEVICE_TABLE(of, t9015_ids); + +static struct platform_driver t9015_driver = { + .driver = { + .name = "t9015-codec", + .of_match_table = of_match_ptr(t9015_ids), + }, + .probe = t9015_probe, +}; + +module_platform_driver(t9015_driver); + +MODULE_DESCRIPTION("ASoC Amlogic T9015 codec driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet@baylibre.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 9841e1da9782..f46d7aca8cf6 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -20,8 +20,8 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int rate = params_rate(params); u32 mclk; int ret; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 295cfffa4646..d4c0f580a565 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -81,6 +81,9 @@ config SND_PXA2XX_SOC_TOSA depends on SND_PXA2XX_SOC && MACH_TOSA depends on MFD_TC6393XB depends on AC97_BUS=n + select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -91,6 +94,9 @@ config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" depends on SND_PXA2XX_SOC && MACH_E740 depends on AC97_BUS=n + select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -101,6 +107,7 @@ config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 depends on AC97_BUS=n + select REGMAP select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -111,7 +118,10 @@ config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 depends on AC97_BUS=n + select REGMAP select SND_SOC_WM9712 + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 help Say Y if you want to add support for SoC audio on the @@ -122,6 +132,9 @@ config SND_PXA2XX_SOC_EM_X270 depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ MACH_CM_X300) depends on AC97_BUS=n + select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -133,6 +146,9 @@ config SND_PXA2XX_SOC_PALM27X depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ MACH_PALMT5 || MACH_PALMTE2) depends on AC97_BUS=n + select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -163,7 +179,10 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE depends on AC97_BUS=n + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 + select REGMAP select SND_PXA_SOC_SSP select SND_SOC_WM9713 help @@ -193,6 +212,9 @@ config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n + select REGMAP + select AC97_BUS_NEW + select AC97_BUS_COMPAT select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 53b1435ced3f..016a91199485 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -44,8 +44,8 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int freq_out, sspa_mclk, sysclk; if (params_rate(params) > 11025) { diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index d81082323fb4..6fbef9a0afa7 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -116,8 +116,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 0139343dbcce..b4da9a9a6521 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -54,8 +54,8 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; /* set the I2S system clock as output */ diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 514e17724fc3..3014e8244ab4 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -12,8 +12,8 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 6483cff5b73d..e4c818f4cd62 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -83,8 +83,8 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int width; int ret = 0; @@ -121,8 +121,8 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; /* set codec DAI configuration */ @@ -358,10 +358,10 @@ static int __init magician_init(void) adapter = i2c_get_adapter(0); if (!adapter) return -ENODEV; - client = i2c_new_device(adapter, i2c_board_info); + client = i2c_new_client_device(adapter, i2c_board_info); i2c_put_adapter(adapter); - if (!client) - return -ENODEV; + if (IS_ERR(client)) + return PTR_ERR(client); ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); if (ret) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 76e054d514a8..bf27b277c01f 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -73,7 +73,7 @@ static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_soc_component *component; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; + component = asoc_rtd_to_codec(rtd, 0)->component; return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event)); } @@ -117,7 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Prepare GPIO8 for rear speaker amplifier */ snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 287b5da739e5..3fe6c4c5a3ab 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -112,7 +112,7 @@ static int mmp_pcm_open(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct platform_device *pdev = to_platform_device(component->dev); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct mmp_dma_data dma_data; struct resource *r; diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index e701637a9ae9..3548a2634a63 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -251,7 +251,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); struct ssp_device *sspa = sspa_priv->sspa; struct snd_dmaengine_dai_dma_data *dma_params; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 59ef04d0467a..287984a564c8 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -90,8 +90,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 5f1c477b5833..9a32bf72127a 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -96,7 +96,7 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index f7babffb7228..6d8174f62935 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -117,8 +117,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index d8f79e2266b1..d5f2961b1a3e 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -61,7 +61,7 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = { static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; /* Headset jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index f9a33cb36f5b..6eee1aefc89a 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -34,8 +34,8 @@ static int z2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 567dc133ea92..447b59b8bd33 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -66,7 +66,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) { if (clk_pout) - snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, + snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, clk_get_rate(pout), 0); return 0; @@ -76,8 +76,8 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 6530d2462a9e..f51b28d1b94d 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,7 +99,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C + depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index ac75838bbfab..2ef090f4af9e 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -33,9 +33,9 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai; struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); int i, rval; @@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dais(rtd, i, codec_dai) { - component = dai->component; + component = codec_dai->component; /* Set default mclk for internal codec */ rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE, SND_SOC_CLOCK_IN); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 94363fd6846a..d55e3ad96716 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -31,8 +31,8 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0; @@ -66,7 +66,7 @@ static struct snd_soc_ops apq8096_ops = { static int apq8096_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* * Codec SLIMBUS configuration diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index b05091c283b7..34f7fd1bab1c 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -55,7 +55,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_runtime->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); struct lpass_variant *v = drvdata->variant; int ret, dma_ch, dir = substream->stream; @@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (substream) { snd_dma_free_pages(&substream->dma_buffer); diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c0d422d0ab94..f6c7cddf08e8 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -41,6 +41,9 @@ #define Q6ASM_DAI_TX 1 #define Q6ASM_DAI_RX 2 +#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) +#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) + enum stream_state { Q6ASM_STREAM_IDLE = 0, Q6ASM_STREAM_STOPPED, @@ -69,6 +72,8 @@ struct q6asm_dai_rtd { }; struct q6asm_dai_data { + struct snd_soc_dai_driver *dais; + int num_dais; long long int sid; }; @@ -250,7 +255,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + 0, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM, prtd->bits_per_sample); @@ -328,7 +333,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0); struct q6asm_dai_rtd *prtd; struct q6asm_dai_data *pdata; struct device *dev = component->dev; @@ -540,7 +545,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) struct snd_soc_pcm_runtime *rtd = stream->private_data; struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct snd_compr_runtime *runtime = stream->runtime; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct q6asm_dai_data *pdata; struct device *dev = c->dev; struct q6asm_dai_rtd *prtd; @@ -627,10 +632,17 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, int dir = stream->direction; struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; + struct q6asm_wma_cfg wma_cfg; + struct q6asm_alac_cfg alac_cfg; + struct q6asm_ape_cfg ape_cfg; + unsigned int wma_v9 = 0; struct device *dev = c->dev; int ret; union snd_codec_options *codec_options; struct snd_dec_flac *flac; + struct snd_dec_wma *wma; + struct snd_dec_alac *alac; + struct snd_dec_ape *ape; codec_options = &(prtd->codec_param.codec.options); @@ -652,7 +664,7 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, prtd->bits_per_sample = 16; if (dir == SND_COMPRESS_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, params->codec.id, - prtd->bits_per_sample); + params->codec.profile, prtd->bits_per_sample); if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); @@ -692,6 +704,126 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, return -EIO; } break; + + case SND_AUDIOCODEC_WMA: + wma = &codec_options->wma_d; + + memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); + + wma_cfg.sample_rate = params->codec.sample_rate; + wma_cfg.num_channels = params->codec.ch_in; + wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; + wma_cfg.block_align = params->codec.align; + wma_cfg.bits_per_sample = prtd->bits_per_sample; + wma_cfg.enc_options = wma->encoder_option; + wma_cfg.adv_enc_options = wma->adv_encoder_option; + wma_cfg.adv_enc_options2 = wma->adv_encoder_option2; + + if (wma_cfg.num_channels == 1) + wma_cfg.channel_mask = 4; /* Mono Center */ + else if (wma_cfg.num_channels == 2) + wma_cfg.channel_mask = 3; /* Stereo FL/FR */ + else + return -EINVAL; + + /* check the codec profile */ + switch (params->codec.profile) { + case SND_AUDIOPROFILE_WMA9: + wma_cfg.fmtag = 0x161; + wma_v9 = 1; + break; + + case SND_AUDIOPROFILE_WMA10: + wma_cfg.fmtag = 0x166; + break; + + case SND_AUDIOPROFILE_WMA9_PRO: + wma_cfg.fmtag = 0x162; + break; + + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + wma_cfg.fmtag = 0x163; + break; + + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + wma_cfg.fmtag = 0x167; + break; + + default: + dev_err(dev, "Unknown WMA profile:%x\n", + params->codec.profile); + return -EIO; + } + + if (wma_v9) + ret = q6asm_stream_media_format_block_wma_v9( + prtd->audio_client, &wma_cfg); + else + ret = q6asm_stream_media_format_block_wma_v10( + prtd->audio_client, &wma_cfg); + if (ret < 0) { + dev_err(dev, "WMA9 CMD failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_ALAC: + memset(&alac_cfg, 0x0, sizeof(alac_cfg)); + alac = &codec_options->alac_d; + + alac_cfg.sample_rate = params->codec.sample_rate; + alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.bit_depth = prtd->bits_per_sample; + alac_cfg.num_channels = params->codec.ch_in; + + alac_cfg.frame_length = alac->frame_length; + alac_cfg.pb = alac->pb; + alac_cfg.mb = alac->mb; + alac_cfg.kb = alac->kb; + alac_cfg.max_run = alac->max_run; + alac_cfg.compatible_version = alac->compatible_version; + alac_cfg.max_frame_bytes = alac->max_frame_bytes; + + switch (params->codec.ch_in) { + case 1: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; + break; + case 2: + alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO; + break; + } + ret = q6asm_stream_media_format_block_alac(prtd->audio_client, + &alac_cfg); + if (ret < 0) { + dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + + case SND_AUDIOCODEC_APE: + memset(&ape_cfg, 0x0, sizeof(ape_cfg)); + ape = &codec_options->ape_d; + + ape_cfg.sample_rate = params->codec.sample_rate; + ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.bits_per_sample = prtd->bits_per_sample; + + ape_cfg.compatible_version = ape->compatible_version; + ape_cfg.compression_level = ape->compression_level; + ape_cfg.format_flags = ape->format_flags; + ape_cfg.blocks_per_frame = ape->blocks_per_frame; + ape_cfg.final_frame_blocks = ape->final_frame_blocks; + ape_cfg.total_frames = ape->total_frames; + ape_cfg.seek_table_present = ape->seek_table_present; + + ret = q6asm_stream_media_format_block_ape(prtd->audio_client, + &ape_cfg); + if (ret < 0) { + dev_err(dev, "APE CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + default: break; } @@ -791,9 +923,12 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 2; + caps->num_codecs = 5; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; + caps->codecs[2] = SND_AUDIOCODEC_WMA; + caps->codecs[3] = SND_AUDIOCODEC_ALAC; + caps->codecs[4] = SND_AUDIOCODEC_APE; return 0; } @@ -889,7 +1024,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = { .compr_ops = &q6asm_dai_compr_ops, }; -static struct snd_soc_dai_driver q6asm_fe_dais[] = { +static struct snd_soc_dai_driver q6asm_fe_dais_template[] = { Q6ASM_FEDAI_DRIVER(1), Q6ASM_FEDAI_DRIVER(2), Q6ASM_FEDAI_DRIVER(3), @@ -903,10 +1038,22 @@ static struct snd_soc_dai_driver q6asm_fe_dais[] = { static int of_q6asm_parse_dai_data(struct device *dev, struct q6asm_dai_data *pdata) { - static struct snd_soc_dai_driver *dai_drv; + struct snd_soc_dai_driver *dai_drv; struct snd_soc_pcm_stream empty_stream; struct device_node *node; - int ret, id, dir; + int ret, id, dir, idx = 0; + + + pdata->num_dais = of_get_child_count(dev->of_node); + if (!pdata->num_dais) { + dev_err(dev, "No dais found in DT\n"); + return -EINVAL; + } + + pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv), + GFP_KERNEL); + if (!pdata->dais) + return -ENOMEM; memset(&empty_stream, 0, sizeof(empty_stream)); @@ -917,7 +1064,8 @@ static int of_q6asm_parse_dai_data(struct device *dev, continue; } - dai_drv = &q6asm_fe_dais[id]; + dai_drv = &pdata->dais[idx++]; + *dai_drv = q6asm_fe_dais_template[id]; ret = of_property_read_u32(node, "direction", &dir); if (ret) @@ -955,11 +1103,12 @@ static int q6asm_dai_probe(struct platform_device *pdev) dev_set_drvdata(dev, pdata); - of_q6asm_parse_dai_data(dev, pdata); + rc = of_q6asm_parse_dai_data(dev, pdata); + if (rc) + return rc; return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component, - q6asm_fe_dais, - ARRAY_SIZE(q6asm_fe_dais)); + pdata->dais, pdata->num_dais); } static const struct of_device_id q6asm_dai_device_id[] = { diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 36e0eab13a98..0e0e8f7a460a 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -39,6 +39,8 @@ #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 #define ASM_MEDIA_FMT_MP3 0x00010BE9 #define ASM_MEDIA_FMT_FLAC 0x00010C16 +#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8 +#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7 #define ASM_DATA_CMD_WRITE_V2 0x00010DAB #define ASM_DATA_CMD_READ_V2 0x00010DAC #define ASM_SESSION_CMD_SUSPEND 0x00010DEC @@ -46,6 +48,8 @@ #define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4 #define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A #define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D +#define ASM_MEDIA_FMT_ALAC 0x00012f31 +#define ASM_MEDIA_FMT_APE 0x00012f32 #define ASM_LEGACY_STREAM_SESSION 0 @@ -104,6 +108,63 @@ struct asm_flac_fmt_blk_v2 { u16 reserved; } __packed; +struct asm_wmastdv9_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 reserved; +} __packed; + +struct asm_wmaprov10_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 fmtag; + u16 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u16 blk_align; + u16 bits_per_sample; + u32 channel_mask; + u16 enc_options; + u16 advanced_enc_options1; + u32 advanced_enc_options2; +} __packed; + +struct asm_alac_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +} __packed; + +struct asm_ape_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -858,7 +919,7 @@ err: * Return: Will be an negative value on error or zero on success */ int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample) + u32 codec_profile, uint16_t bits_per_sample) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; @@ -894,6 +955,30 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, case SND_AUDIOCODEC_FLAC: open->dec_fmt_id = ASM_MEDIA_FMT_FLAC; break; + case SND_AUDIOCODEC_WMA: + switch (codec_profile) { + case SND_AUDIOPROFILE_WMA9: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9; + break; + case SND_AUDIOPROFILE_WMA10: + case SND_AUDIOPROFILE_WMA9_PRO: + case SND_AUDIOPROFILE_WMA9_LOSSLESS: + case SND_AUDIOPROFILE_WMA10_LOSSLESS: + open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10; + break; + default: + dev_err(ac->dev, "Invalid codec profile 0x%x\n", + codec_profile); + rc = -EINVAL; + goto err; + } + break; + case SND_AUDIOCODEC_ALAC: + open->dec_fmt_id = ASM_MEDIA_FMT_ALAC; + break; + case SND_AUDIOCODEC_APE: + open->dec_fmt_id = ASM_MEDIA_FMT_APE; + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1075,6 +1160,162 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, return rc; } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac); + +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmastdv9_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->reserved = 0; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9); + +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg) +{ + struct asm_wmaprov10_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->fmtag = cfg->fmtag; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->bytes_per_sec = cfg->bytes_per_sec; + fmt->blk_align = cfg->block_align; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->channel_mask = cfg->channel_mask; + fmt->enc_options = cfg->enc_options; + fmt->advanced_enc_options1 = cfg->adv_enc_options; + fmt->advanced_enc_options2 = cfg->adv_enc_options2; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10); + +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg) +{ + struct asm_alac_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->frame_length = cfg->frame_length; + fmt->compatible_version = cfg->compatible_version; + fmt->bit_depth = cfg->bit_depth; + fmt->num_channels = cfg->num_channels; + fmt->max_run = cfg->max_run; + fmt->max_frame_bytes = cfg->max_frame_bytes; + fmt->avg_bit_rate = cfg->avg_bit_rate; + fmt->sample_rate = cfg->sample_rate; + fmt->channel_layout_tag = cfg->channel_layout_tag; + fmt->pb = cfg->pb; + fmt->mb = cfg->mb; + fmt->kb = cfg->kb; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac); + +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg) +{ + struct asm_ape_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + + fmt->compatible_version = cfg->compatible_version; + fmt->compression_level = cfg->compression_level; + fmt->format_flags = cfg->format_flags; + fmt->blocks_per_frame = cfg->blocks_per_frame; + fmt->final_frame_blocks = cfg->final_frame_blocks; + fmt->total_frames = cfg->total_frames; + fmt->bits_per_sample = cfg->bits_per_sample; + fmt->num_channels = cfg->num_channels; + fmt->sample_rate = cfg->sample_rate; + fmt->seek_table_present = cfg->seek_table_present; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 6764f55f7078..38a207d6cd95 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -45,6 +45,47 @@ struct q6asm_flac_cfg { u16 md5_sum; }; +struct q6asm_wma_cfg { + u32 fmtag; + u32 num_channels; + u32 sample_rate; + u32 bytes_per_sec; + u32 block_align; + u32 bits_per_sample; + u32 channel_mask; + u32 enc_options; + u32 adv_enc_options; + u32 adv_enc_options2; +}; + +struct q6asm_alac_cfg { + u32 frame_length; + u8 compatible_version; + u8 bit_depth; + u8 pb; + u8 mb; + u8 kb; + u8 num_channels; + u16 max_run; + u32 max_frame_bytes; + u32 avg_bit_rate; + u32 sample_rate; + u32 channel_layout_tag; +}; + +struct q6asm_ape_cfg { + u16 compatible_version; + u16 compression_level; + u32 format_flags; + u32 blocks_per_frame; + u32 final_frame_blocks; + u32 total_frames; + u16 bits_per_sample; + u16 num_channels; + u32 sample_rate; + u32 seek_table_present; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -55,7 +96,7 @@ void q6asm_audio_client_free(struct audio_client *ac); int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); int q6asm_open_write(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample); + u32 codec_profile, uint16_t bits_per_sample); int q6asm_open_read(struct audio_client *ac, uint32_t format, uint16_t bits_per_sample); @@ -69,6 +110,14 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, uint16_t bits_per_sample); int q6asm_stream_media_format_block_flac(struct audio_client *ac, struct q6asm_flac_cfg *cfg); +int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + struct q6asm_wma_cfg *cfg); +int q6asm_stream_media_format_block_alac(struct audio_client *ac, + struct q6asm_alac_cfg *cfg); +int q6asm_stream_media_format_block_ape(struct audio_client *ac, + struct q6asm_ape_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 20724102e85a..46e50612b92c 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -918,25 +918,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL7", NULL, "MultiMedia7 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, - - {"MM_DL1", NULL, "MultiMedia1 Playback" }, - {"MM_DL2", NULL, "MultiMedia2 Playback" }, - {"MM_DL3", NULL, "MultiMedia3 Playback" }, - {"MM_DL4", NULL, "MultiMedia4 Playback" }, - {"MM_DL5", NULL, "MultiMedia5 Playback" }, - {"MM_DL6", NULL, "MultiMedia6 Playback" }, - {"MM_DL7", NULL, "MultiMedia7 Playback" }, - {"MM_DL8", NULL, "MultiMedia8 Playback" }, - - {"MultiMedia1 Capture", NULL, "MM_UL1"}, - {"MultiMedia2 Capture", NULL, "MM_UL2"}, - {"MultiMedia3 Capture", NULL, "MM_UL3"}, - {"MultiMedia4 Capture", NULL, "MM_UL4"}, - {"MultiMedia5 Capture", NULL, "MM_UL5"}, - {"MultiMedia6 Capture", NULL, "MM_UL6"}, - {"MultiMedia7 Capture", NULL, "MM_UL7"}, - {"MultiMedia8 Capture", NULL, "MM_UL8"}, - }; static int routing_hw_params(struct snd_soc_component *component, @@ -945,7 +926,7 @@ static int routing_hw_params(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct msm_routing_data *data = dev_get_drvdata(component->dev); - unsigned int be_id = rtd->cpu_dai->id; + unsigned int be_id = asoc_rtd_to_cpu(rtd, 0)->id; struct session_data *session; int path_type; diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 3b5547a27aad..b2de65c7f95c 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -11,6 +11,7 @@ #include <sound/pcm_params.h> #include <sound/jack.h> #include <sound/soc.h> +#include <linux/soundwire/sdw.h> #include <uapi/linux/input-event-codes.h> #include "common.h" #include "qdsp6/q6afe.h" @@ -31,10 +32,12 @@ struct sdm845_snd_data { struct snd_soc_jack jack; bool jack_setup; + bool stream_prepared[SLIM_MAX_RX_PORTS]; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; uint32_t sec_mi2s_clk_count; uint32_t quat_tdm_clk_count; + struct sdw_stream_runtime *sruntime[SLIM_MAX_RX_PORTS]; }; static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28}; @@ -43,14 +46,21 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai_link = rtd->dai_link; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai; + struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; + struct sdw_stream_runtime *sruntime; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; - for (i = 0 ; i < dai_link->num_codecs; i++) { - ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dais(rtd, i, codec_dai) { + sruntime = snd_soc_dai_get_sdw_stream(codec_dai, + substream->stream); + if (sruntime != ERR_PTR(-ENOTSUPP)) + pdata->sruntime[cpu_dai->id] = sruntime; + + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -76,7 +86,8 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai; int ret = 0, j; int channels, slot_width; @@ -125,8 +136,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, } } - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { ret = snd_soc_dai_set_tdm_slot( @@ -161,8 +171,8 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret = 0; switch (cpu_dai->id) { @@ -210,11 +220,10 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; - struct snd_soc_dai_link *dai_link = rtd->dai_link; /* * Codec SLIMBUS configuration * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13 @@ -266,8 +275,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: - for (i = 0 ; i < dai_link->num_codecs; i++) { - rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dais(rtd, i, codec_dai) { + rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, ARRAY_SIZE(rx_ch), @@ -275,7 +284,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) if (rval != 0 && rval != -ENOTSUPP) return rval; - snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0, + snd_soc_dai_set_sysclk(codec_dai, 0, WCD934X_DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); } @@ -295,8 +304,8 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int j; int ret; @@ -345,8 +354,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; - for (j = 0; j < rtd->num_codecs; j++) { - codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { @@ -386,7 +394,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct sdm845_snd_data *data = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { case PRIMARY_MI2S_RX: @@ -427,8 +435,65 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) } } +static int sdm845_snd_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + int ret; + + if (!sruntime) + return 0; + + if (data->stream_prepared[cpu_dai->id]) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + data->stream_prepared[cpu_dai->id] = false; + } + + ret = sdw_prepare_stream(sruntime); + if (ret) + return ret; + + /** + * NOTE: there is a strict hw requirement about the ordering of port + * enables and actual WSA881x PA enable. PA enable should only happen + * after soundwire ports are enabled if not DC on the line is + * accumulated resulting in Click/Pop Noise + * PA enable/mute are handled as part of codec DAPM and digital mute. + */ + + ret = sdw_enable_stream(sruntime); + if (ret) { + sdw_deprepare_stream(sruntime); + return ret; + } + data->stream_prepared[cpu_dai->id] = true; + + return ret; +} + +static int sdm845_snd_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sdm845_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + if (sruntime && data->stream_prepared[cpu_dai->id]) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + data->stream_prepared[cpu_dai->id] = false; + } + + return 0; +} + static const struct snd_soc_ops sdm845_be_ops = { .hw_params = sdm845_snd_hw_params, + .hw_free = sdm845_snd_hw_free, + .prepare = sdm845_snd_prepare, .startup = sdm845_snd_startup, .shutdown = sdm845_snd_shutdown, }; diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index e6666e597265..3a6e18709b9e 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -39,7 +39,7 @@ static int storm_ops_hw_params(struct snd_pcm_substream *substream, */ sysclk_freq = rate * bitwidth * 2 * STORM_SYSCLK_MULT; - ret = snd_soc_dai_set_sysclk(soc_runtime->cpu_dai, 0, sysclk_freq, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(soc_runtime, 0), 0, sysclk_freq, 0); if (ret) { dev_err(card->dev, "error setting sysclk to %u: %d\n", sysclk_freq, ret); diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 767700c34ee2..01078155a914 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -67,8 +67,8 @@ static int rk_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index d951100bf770..f45e5aaa4b30 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -57,7 +57,7 @@ static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substrea mclk = params_rate(params) * SOUND_FS; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0); if (ret) { dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", __func__, mclk, ret); @@ -71,8 +71,8 @@ static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int mclk; int ret; @@ -103,8 +103,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk, ret; /* in bypass mode, the mclk has to be one of the frequencies below */ @@ -153,8 +153,8 @@ static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dais[0]->component; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; /* We need default MCLK and PLL settings for the accessory detection */ @@ -206,7 +206,7 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, mclk = params_rate(params) * SOUND_FS; - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), 0, mclk, 0); if (ret) { dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", __func__, mclk, ret); diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 60930fa85aa4..1f527d3763ce 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -146,8 +146,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { @@ -227,7 +227,7 @@ static struct snd_soc_jack rk_hdmi_jack; static int rk_hdmi_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; - struct snd_soc_component *component = runtime->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; int ret; /* enable jack detection */ diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 26b67b245484..0617ccf4e42c 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -56,8 +56,8 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int mclk; switch (params_rate(params)) { @@ -113,7 +113,7 @@ static int rk_init(struct snd_soc_pcm_runtime *runtime) return ret; } - return rt5645_set_jack_detect(runtime->codec_dai->component, + return rt5645_set_jack_detect(asoc_rtd_to_codec(runtime, 0)->component, &headset_jack, &headset_jack, &headset_jack); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1a0b163ca47b..112911dc271b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -204,7 +204,7 @@ config SND_SOC_ARNDALE config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index d64602950cbd..c81ece78e036 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -21,8 +21,8 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int rfs, ret; unsigned long rclk; @@ -56,7 +56,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int rfs, rclk; /* Ensure AIF1CLK is >= 3 MHz for optimal performance */ @@ -174,7 +174,9 @@ static int arndale_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(card->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); goto err_put_of_nodes; } return 0; diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 5de633497f83..8b83f39c3ac9 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -60,7 +60,7 @@ static int bells_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); component = codec_dai->component; if (dapm->dev != codec_dai->dev) @@ -106,7 +106,7 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); component = codec_dai->component; if (dapm->dev != codec_dai->dev) @@ -152,11 +152,11 @@ static int bells_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_AP_DSP]); - wm0010 = rtd->codec_dai->component; + wm0010 = asoc_rtd_to_codec(rtd, 0)->component; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_DSP_CODEC]); - component = rtd->codec_dai->component; - aif1_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_component_set_sysclk(component, ARIZONA_CLK_SYSCLK, ARIZONA_CLK_SRC_FLL1, @@ -195,7 +195,7 @@ static int bells_late_probe(struct snd_soc_card *card) } rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_CP]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret != 0) { @@ -207,8 +207,8 @@ static int bells_late_probe(struct snd_soc_card *card) return 0; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[DAI_CODEC_SUB]); - aif3_dai = rtd->cpu_dai; - wm9081_dai = rtd->codec_dai; + aif3_dai = asoc_rtd_to_cpu(rtd, 0); + wm9081_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret != 0) { diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index a95c34e53a2b..9139a1e7e200 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -68,7 +68,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int div; int ret; unsigned int rate = params_rate(params); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a57bb989a0ef..f86e3028b402 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -932,7 +932,7 @@ static int i2s_trigger(struct snd_pcm_substream *substream, struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct i2s_dai *i2s = to_info(rtd->cpu_dai); + struct i2s_dai *i2s = to_info(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; switch (cmd) { diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 949d2e029962..30899016cf08 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -33,8 +33,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct s3c_i2sv2_rate_calc div; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 59904f44118b..f4375c49f7f4 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,7 +23,7 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - aif1_dai = rtd->codec_dai; + aif1_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != aif1_dai->dev) return 0; @@ -70,7 +70,7 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - aif1_dai = rtd->codec_dai; + aif1_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != aif1_dai->dev) return 0; @@ -105,7 +105,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; sample_rate = params_rate(params); @@ -181,7 +181,7 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -264,11 +264,11 @@ static int littlemill_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; - aif1_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - aif2_dai = rtd->cpu_dai; + aif2_dai = asoc_rtd_to_cpu(rtd, 0); ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); @@ -325,7 +325,7 @@ static int littlemill_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 098eefc764db..998d10cf8c94 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -32,7 +32,7 @@ static struct snd_soc_jack_pin lowland_headset_pins[] = { static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; ret = snd_soc_component_set_sysclk(component, WM5100_CLK_SYSCLK, @@ -65,7 +65,7 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) static int lowland_wm9081_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; snd_soc_dapm_nc_pin(&rtd->card->dapm, "LINEOUT"); @@ -183,7 +183,7 @@ static int lowland_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 1339e41e9860..b7ce1da854ce 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -26,8 +26,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -100,7 +100,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); @@ -118,7 +118,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -155,7 +155,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index f0f5fa9c27d3..6eda5af989fe 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -98,7 +98,7 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; if (rtd->num_codecs > 1) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[1]; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 1); ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk_freq, SND_SOC_CLOCK_IN); @@ -311,7 +311,9 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "snd_soc_register_card() failed: %d\n", + ret); goto err_put_clk_i2s; } diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index f6e67d0e7882..a5b1a12b3496 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -212,7 +212,7 @@ static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; dev_dbg(pcm->dev, "Entered %s\n", __func__); @@ -256,7 +256,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *socdai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 4b247e91ae5b..3afe63c0923e 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -149,7 +149,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int div; int ret; unsigned int rate = params_rate(params); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 593be1b668d6..358887848293 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -380,7 +380,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_i2sv2_info *i2s = to_info(rtd->cpu_dai); + struct s3c_i2sv2_info *i2s = to_info(asoc_rtd_to_cpu(rtd, 0)); int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index 4543705b8d87..fd2a4da086f3 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -160,8 +160,8 @@ static int simtec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(codec_dai, 0, diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 55d2a802a6cb..abb5c4713c53 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -51,7 +51,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c24xx_uda134x *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; mutex_lock(&priv->clk_lock); @@ -119,8 +119,8 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret = 0; int clk_source, fs_mode; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index fab3db9fdb98..36bef136d57f 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -25,8 +25,8 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int clk = 0; int ret; diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index 4baef84d29ee..776a270261bf 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -101,7 +101,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned long pll_out, rclk_rate; int ret, ratio; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index d096ff912260..02074c34a2b2 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -23,7 +23,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pll_out; int rfs, ret; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 28f8be000aa1..a9f345f19a8a 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -45,7 +45,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int pll_out; int ret; @@ -178,7 +178,7 @@ static int smdk_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e3dc7320c62..746930dde5d7 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -44,8 +44,8 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned long mclk_freq; int rfs, ret; @@ -118,7 +118,7 @@ static int snd_smdk_probe(struct platform_device *pdev) smdk_pcm.dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); return ret; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index f075aae9561a..40c5de8df0ff 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -110,9 +110,9 @@ static int snow_late_probe(struct snd_soc_card *card) /* In the multi-codec case codec_dais 0 is MAX98095 and 1 is HDMI. */ if (rtd->num_codecs > 1) - codec_dai = rtd->codec_dais[0]; + codec_dai = asoc_rtd_to_codec(rtd, 0); else - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set the MCLK rate for the codec */ return snd_soc_dai_set_sysclk(codec_dai, 0, @@ -216,7 +216,9 @@ static int snow_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card failed (%d)\n", ret); return ret; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 1a9f08a50394..759fc6644329 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -142,7 +142,7 @@ static int spdif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); unsigned long flags; dev_dbg(spdif->dev, "Entered %s\n", __func__); @@ -178,7 +178,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *socdai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = spdif->regs; struct snd_dmaengine_dai_dma_data *dma_data; u32 con, clkcon, cstas; @@ -194,7 +194,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + snd_soc_dai_set_dma_data(asoc_rtd_to_cpu(rtd, 0), substream, dma_data); spin_lock_irqsave(&spdif->lock, flags); @@ -280,7 +280,7 @@ static void spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct samsung_spdif_info *spdif = to_info(rtd->cpu_dai); + struct samsung_spdif_info *spdif = to_info(asoc_rtd_to_cpu(rtd, 0)); void __iomem *regs = spdif->regs; u32 con, clkcon; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index ea0d1ec67f01..f5f6ba00d073 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -61,7 +61,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[1]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -131,7 +131,7 @@ static void speyside_set_polarity(struct snd_soc_component *component, static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); int ret; ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0); @@ -143,7 +143,7 @@ static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd) static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = dai->component; int ret; @@ -330,7 +330,7 @@ static int speyside_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 10ff14b856f2..6dfd540e2d74 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -93,7 +93,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct tm2_machine_priv *priv = snd_soc_card_get_drvdata(rtd->card); switch (params_rate(params)) { @@ -134,7 +134,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; unsigned int asyncclk_rate; int ret; @@ -188,7 +188,7 @@ static int tm2_aif2_hw_params(struct snd_pcm_substream *substream, static int tm2_aif2_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; int ret; /* disable FLL2 */ @@ -209,7 +209,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); unsigned int bfs; int bitwidth, ret; @@ -284,7 +284,7 @@ static int tm2_set_bias_level(struct snd_soc_card *card, rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - if (dapm->dev != rtd->codec_dai->dev) + if (dapm->dev != asoc_rtd_to_codec(rtd, 0)->dev) return 0; switch (level) { @@ -315,8 +315,8 @@ static int tm2_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF1]); - aif1_dai = rtd->codec_dai; - priv->component = rtd->codec_dai->component; + aif1_dai = asoc_rtd_to_codec(rtd, 0); + priv->component = asoc_rtd_to_codec(rtd, 0)->component; ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret < 0) { @@ -325,7 +325,7 @@ static int tm2_late_probe(struct snd_soc_card *card) } rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[TM2_DAI_AIF2]); - aif2_dai = rtd->codec_dai; + aif2_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret < 0) { @@ -611,7 +611,8 @@ static int tm2_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "Failed to register card: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to register card: %d\n", ret); goto dai_node_put; } diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index fdce28cc26c4..c962d2c2a7f7 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -23,7 +23,7 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -66,7 +66,7 @@ static int tobermory_set_bias_level_post(struct snd_soc_card *card, int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = rtd->codec_dai; + codec_dai = asoc_rtd_to_codec(rtd, 0); if (dapm->dev != codec_dai->dev) return 0; @@ -181,8 +181,8 @@ static int tobermory_late_probe(struct snd_soc_card *card) int ret; rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - component = rtd->codec_dai->component; - codec_dai = rtd->codec_dai; + component = asoc_rtd_to_codec(rtd, 0)->component; + codec_dai = asoc_rtd_to_codec(rtd, 0); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); @@ -229,7 +229,7 @@ static int tobermory_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index eee1a1e994cb..a35de78f14a9 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -119,7 +119,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int ret, dmairq; @@ -132,7 +132,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); if (unlikely(ret)) { pr_debug("audio unit %d irqs already taken!\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EBUSY; } (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); @@ -141,7 +141,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); if (unlikely(ret)) { pr_debug("audio unit %d irqs already taken!\n", - rtd->cpu_dai->id); + asoc_rtd_to_cpu(rtd, 0)->id); return -EBUSY; } (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); @@ -153,7 +153,7 @@ static int camelot_pcm_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int dmairq; @@ -175,7 +175,7 @@ static int camelot_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *hw_params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; int ret; @@ -194,7 +194,7 @@ static int camelot_prepare(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); @@ -242,7 +242,7 @@ static int camelot_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; switch (cmd) { @@ -270,7 +270,7 @@ static snd_pcm_uframes_t camelot_pos(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; + struct camelot_pcm *cam = &cam_pcm_data[asoc_rtd_to_cpu(rtd, 0)->id]; int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; unsigned long pos; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4b35ef402604..1c3c4fdc9bef 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -408,7 +408,7 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai) @@ -1938,8 +1938,7 @@ static int fsi_probe(struct platform_device *pdev) if (!master) return -ENOMEM; - master->base = devm_ioremap(&pdev->dev, - res->start, resource_size(res)); + master->base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (!master->base) { dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); return -ENXIO; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 991557e25eba..d5702fbf176b 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -46,7 +46,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int ret; unsigned int rate = params_rate(params); @@ -67,7 +67,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, clk_set_rate(&siumckb_clk, codec_freq); dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); - ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, SIU_CLKB_EXT, + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), SIU_CLKB_EXT, codec_freq / 2, SND_SOC_CLOCK_IN); if (!ret) @@ -79,7 +79,7 @@ static int migor_hw_params(struct snd_pcm_substream *substream, static int migor_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); if (use_count) { use_count--; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0bfcb77e5f65..4349f2fb823f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -696,7 +696,7 @@ struct snd_soc_dai *rsnd_substream_to_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - return rtd->cpu_dai; + return asoc_rtd_to_cpu(rtd, 0); } static diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 392a1c5b15d3..50062eb79adb 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -810,9 +810,10 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) int playback = 0, capture = 0; int i; - if (rtd->num_codecs > 1) { + if (rtd->num_cpus > 1 || + rtd->num_codecs > 1) { dev_err(rtd->card->dev, - "Compress ASoC: Multicodec not supported\n"); + "Compress ASoC: Multi CPU/Codec not supported\n"); return -EINVAL; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..843b8b1c89d4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", + codec_dai->stream_active[playback] ? "active" : "inactive", rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ if (rtd->pop_wait == 1) { rtd->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + snd_soc_dapm_stream_event(rtd, playback, SND_SOC_DAPM_STREAM_STOP); } @@ -431,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_component *component; struct device *dev; int ret; + int stream; /* * for rtd->dev @@ -465,23 +467,31 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; INIT_LIST_HEAD(&rtd->list); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); + for_each_pcm_streams(stream) { + INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients); + } dev_set_drvdata(dev, rtd); INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* - * for rtd->codec_dais + * for rtd->dais */ - rtd->codec_dais = devm_kcalloc(dev, dai_link->num_codecs, + rtd->dais = devm_kcalloc(dev, dai_link->num_cpus + dai_link->num_codecs, sizeof(struct snd_soc_dai *), GFP_KERNEL); - if (!rtd->codec_dais) + if (!rtd->dais) goto free_rtd; /* + * dais = [][][][][][][][][][][][][][][][][][] + * ^cpu_dais ^codec_dais + * |--- num_cpus ---|--- num_codecs --| + */ + rtd->cpu_dais = &rtd->dais[0]; + rtd->codec_dais = &rtd->dais[dai_link->num_cpus]; + + /* * rtd remaining settings */ rtd->card = card; @@ -514,6 +524,7 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_component *component; struct snd_soc_pcm_runtime *rtd; + int playback = SNDRV_PCM_STREAM_PLAYBACK; int i; /* If the card is not initialized yet there is nothing to do */ @@ -536,10 +547,9 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue; - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 1, - SNDRV_PCM_STREAM_PLAYBACK); + for_each_rtd_codec_dais(rtd, i, dai) { + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 1, playback); } } @@ -558,17 +568,14 @@ int snd_soc_suspend(struct device *dev) snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_SUSPEND); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -664,30 +671,27 @@ static void soc_resume_deferred(struct work_struct *work) } for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ for_each_card_rtds(card, rtd) { struct snd_soc_dai *dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; if (rtd->dai_link->ignore_suspend) continue; - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 0, - SNDRV_PCM_STREAM_PLAYBACK); + for_each_rtd_codec_dais(rtd, i, dai) { + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 0, playback); } } @@ -837,7 +841,7 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { int i; - struct snd_soc_dai_link_component *codec, *platform; + struct snd_soc_dai_link_component *cpu, *codec, *platform; for_each_link_codecs(link, i, codec) { /* @@ -886,44 +890,38 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, return -EPROBE_DEFER; } - /* FIXME */ - if (link->num_cpus > 1) { - dev_err(card->dev, - "ASoC: multi cpu is not yet supported %s\n", - link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpus->name && link->cpus->of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } + for_each_link_cpus(link, i, cpu) { + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (cpu->name && cpu->of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } - /* - * Defer card registration if cpu dai component is not added to - * component list. - */ - if ((link->cpus->of_node || link->cpus->name) && - !soc_find_component(link->cpus)) - return -EPROBE_DEFER; + /* + * Defer card registration if cpu dai component is not added to + * component list. + */ + if ((cpu->of_node || cpu->name) && + !soc_find_component(cpu)) + return -EPROBE_DEFER; - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpus->dai_name && - !(link->cpus->name || link->cpus->of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!cpu->dai_name && + !(cpu->name || cpu->of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } } return 0; @@ -966,7 +964,7 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai_link_component *codec, *platform; + struct snd_soc_dai_link_component *codec, *platform, *cpu; struct snd_soc_component *component; int i, ret; @@ -991,14 +989,19 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, if (!rtd) return -ENOMEM; - /* FIXME: we need multi CPU support in the future */ - rtd->cpu_dai = snd_soc_find_dai(dai_link->cpus); - if (!rtd->cpu_dai) { - dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", - dai_link->cpus->dai_name); - goto _err_defer; + rtd->num_cpus = dai_link->num_cpus; + for_each_link_cpus(dai_link, i, cpu) { + rtd->cpu_dais[i] = snd_soc_find_dai(cpu); + if (!rtd->cpu_dais[i]) { + dev_info(card->dev, "ASoC: CPU DAI %s not registered\n", + cpu->dai_name); + goto _err_defer; + } + snd_soc_rtd_add_component(rtd, rtd->cpu_dais[i]->component); } - snd_soc_rtd_add_component(rtd, rtd->cpu_dai->component); + + /* Single cpu links expect cpu and cpu_dai in runtime data */ + rtd->cpu_dai = rtd->cpu_dais[0]; /* Find CODEC from registered CODECs */ rtd->num_codecs = dai_link->num_codecs; @@ -1034,20 +1037,20 @@ _err_defer: } EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime); -static int soc_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, - struct snd_soc_pcm_runtime *rtd) +static int soc_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai; int i, ret = 0; - for (i = 0; i < num_dais; ++i) { - struct snd_soc_dai_driver *drv = dais[i]->driver; + for_each_rtd_dais(rtd, i, dai) { + struct snd_soc_dai_driver *drv = dai->driver; if (drv->pcm_new) - ret = drv->pcm_new(rtd, dais[i]); + ret = drv->pcm_new(rtd, dai); if (ret < 0) { - dev_err(dais[i]->dev, + dev_err(dai->dev, "ASoC: Failed to bind %s with pcm device\n", - dais[i]->name); + dai->name); return ret; } } @@ -1118,12 +1121,8 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, dai_link->stream_name, ret); return ret; } - ret = soc_dai_pcm_new(&cpu_dai, 1, rtd); - if (ret < 0) - return ret; - ret = soc_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - return ret; + + return soc_dai_pcm_new(rtd); } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1256,8 +1255,18 @@ static int soc_probe_component(struct snd_soc_card *card, ret = snd_soc_dapm_add_routes(dapm, component->driver->dapm_routes, component->driver->num_dapm_routes); - if (ret < 0) - goto err_probe; + if (ret < 0) { + if (card->disable_route_checks) { + dev_info(card->dev, + "%s: disable_route_checks set, ignoring errors on add_routes\n", + __func__); + } else { + dev_err(card->dev, + "%s: snd_soc_dapm_add_routes failed: %d\n", + __func__, ret); + goto err_probe; + } + } /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); @@ -1309,24 +1318,22 @@ static int soc_probe_dai(struct snd_soc_dai *dai, int order) static void soc_remove_link_dais(struct snd_soc_card *card) { int i; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; struct snd_soc_pcm_runtime *rtd; int order; for_each_comp_order(order) { for_each_card_rtds(card, rtd) { - /* remove the CODEC DAI */ - for_each_rtd_codec_dai(rtd, i, codec_dai) - soc_remove_dai(codec_dai, order); - - soc_remove_dai(rtd->cpu_dai, order); + /* remove DAIs */ + for_each_rtd_dais(rtd, i, dai) + soc_remove_dai(dai, order); } } } static int soc_probe_link_dais(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; struct snd_soc_pcm_runtime *rtd; int i, order, ret; @@ -1337,13 +1344,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card) "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); - ret = soc_probe_dai(rtd->cpu_dai, order); - if (ret) - return ret; - - /* probe the CODEC DAI */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = soc_probe_dai(codec_dai, order); + /* probe the CPU DAI */ + for_each_rtd_dais(rtd, i, dai) { + ret = soc_probe_dai(dai, order); if (ret) return ret; } @@ -1471,12 +1474,13 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, unsigned int dai_fmt) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + unsigned int inv_dai_fmt; unsigned int i; int ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); if (ret != 0 && ret != -ENOTSUPP) { dev_warn(codec_dai->dev, @@ -1489,33 +1493,33 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, * Flip the polarity for the "CPU" end of a CODEC<->CODEC link * the component which has non_legacy_dai_naming is Codec */ - if (cpu_dai->component->driver->non_legacy_dai_naming) { - unsigned int inv_dai_fmt; - - inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; - switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case SND_SOC_DAIFMT_CBM_CFS: - inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case SND_SOC_DAIFMT_CBS_CFS: - inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - dai_fmt = inv_dai_fmt; + inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; } + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + unsigned int fmt = dai_fmt; - ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) { - dev_warn(cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", ret); - return ret; + if (cpu_dai->component->driver->non_legacy_dai_naming) + fmt = inv_dai_fmt; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(cpu_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } } return 0; @@ -1938,8 +1942,18 @@ static int snd_soc_bind_card(struct snd_soc_card *card) ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); - if (ret < 0) - goto probe_end; + if (ret < 0) { + if (card->disable_route_checks) { + dev_info(card->dev, + "%s: disable_route_checks set, ignoring errors on add_routes\n", + __func__); + } else { + dev_err(card->dev, + "%s: snd_soc_dapm_add_routes failed: %d\n", + __func__, ret); + goto probe_end; + } + } ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); @@ -3102,6 +3116,14 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, *dai_name = dai->driver->name; if (!*dai_name) *dai_name = pos->name; + } else if (ret) { + /* + * if another error than ENOTSUPP is returned go on and + * check if another component is provided with the same + * node. This may happen if a device provides several + * components + */ + continue; } break; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 51031e330179..19142f6e533c 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -295,17 +295,24 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; - if (dai->driver->ops->startup) + if (!dai->started && + dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); + if (ret == 0) + dai->started = 1; + return ret; } void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream) { - if (dai->driver->ops->shutdown) + if (dai->started && + dai->driver->ops->shutdown) dai->driver->ops->shutdown(substream, dai); + + dai->started = 0; } int snd_soc_dai_prepare(struct snd_soc_dai *dai, @@ -383,12 +390,7 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai, */ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) { - struct snd_soc_pcm_stream *stream; - - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - stream = &dai->driver->playback; - else - stream = &dai->driver->capture; + struct snd_soc_pcm_stream *stream = snd_soc_dai_get_pcm_stream(dai, dir); /* If the codec specifies any channels at all, it supports the stream */ return stream->channels_min; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..04da7928c873 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -302,7 +302,7 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { if (w->is_ep) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) @@ -589,7 +589,7 @@ static void dapm_reset(struct snd_soc_card *card) memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { w->new_power = w->power; w->power_checked = false; } @@ -833,7 +833,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, *kcontrol = NULL; - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (w == kcontrolw || w->dapm != kcontrolw->dapm) continue; for (i = 0; i < w->num_kcontrols; i++) { @@ -1105,6 +1105,11 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +static void dapm_widget_list_free(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, struct list_head *widgets) { @@ -1310,6 +1315,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, return paths; } +void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list) +{ + dapm_widget_list_free(list); +} + /* * Handler for regulator supply widget. */ @@ -1706,9 +1716,8 @@ static void dapm_seq_run(struct snd_soc_card *card, i, cur_subseq); } - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) soc_dapm_async_complete(d); - } } static void dapm_widget_update(struct snd_soc_card *card) @@ -1724,9 +1733,7 @@ static void dapm_widget_update(struct snd_soc_card *card) wlist = dapm_kcontrol_get_wlist(update->kcontrol); - for (wi = 0; wi < wlist->num_widgets; wi++) { - w = wlist->widgets[wi]; - + for_each_dapm_widgets(wlist, wi, w) { if (w->event && (w->event_flags & SND_SOC_DAPM_PRE_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_PRE_REG); if (ret != 0) @@ -1753,9 +1760,7 @@ static void dapm_widget_update(struct snd_soc_card *card) w->name, ret); } - for (wi = 0; wi < wlist->num_widgets; wi++) { - w = wlist->widgets[wi]; - + for_each_dapm_widgets(wlist, wi, w) { if (w->event && (w->event_flags & SND_SOC_DAPM_POST_REG)) { ret = w->event(w, update->kcontrol, SND_SOC_DAPM_POST_REG); if (ret != 0) @@ -1943,7 +1948,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_start(card); - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (dapm_idle_bias_off(d)) d->target_bias_level = SND_SOC_BIAS_OFF; else @@ -1962,7 +1967,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_power_one_widget(w, &up_list, &down_list); } - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { switch (w->id) { case snd_soc_dapm_pre: case snd_soc_dapm_post: @@ -2007,10 +2012,10 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) * they're not ground referenced. */ bias = SND_SOC_BIAS_OFF; - list_for_each_entry(d, &card->dapm_list, list) + for_each_card_dapms(card, d) if (d->target_bias_level > bias) bias = d->target_bias_level; - list_for_each_entry(d, &card->dapm_list, list) + for_each_card_dapms(card, d) if (!dapm_idle_bias_off(d)) d->target_bias_level = bias; @@ -2019,7 +2024,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run card bias changes at first */ dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); @@ -2043,7 +2048,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); @@ -2053,7 +2058,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_post_sequence_async(&card->dapm, 0); /* do we need to notify any clients that DAPM event is complete */ - list_for_each_entry(d, &card->dapm_list, list) { + for_each_card_dapms(card, d) { if (!d->component) continue; @@ -2286,7 +2291,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); @@ -2351,7 +2356,7 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); @@ -2371,7 +2376,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, if (!cmpnt->card) return 0; - list_for_each_entry(w, &cmpnt->card->widgets, list) { + for_each_card_widgets(cmpnt->card, w) { if (w->dapm != dapm) continue; @@ -2431,7 +2436,7 @@ static ssize_t dapm_widget_show(struct device *dev, mutex_lock(&rtd->card->dapm_mutex); - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_soc_component *cmpnt = codec_dai->component; count += dapm_widget_show_component(cmpnt, buf + count); @@ -2491,7 +2496,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w, *next_w; - list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { + for_each_card_widgets_safe(dapm->card, w, next_w) { if (w->dapm != dapm) continue; snd_soc_dapm_free_widget(w); @@ -2506,7 +2511,7 @@ static struct snd_soc_dapm_widget *dapm_find_widget( struct snd_soc_dapm_widget *w; struct snd_soc_dapm_widget *fallback = NULL; - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (!strcmp(w->name, pin)) { if (w->dapm == dapm) return w; @@ -2624,10 +2629,7 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w; int ret; - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, dir); if (!w) return 0; @@ -2908,7 +2910,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, * find src and dest widgets over all widgets but favor a widget from * current DAPM context */ - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (!wsink && !(strcmp(w->name, sink))) { wtsink = w; if (w->dapm == dapm) { @@ -3187,7 +3189,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); - list_for_each_entry(w, &card->widgets, list) + for_each_card_widgets(card, w) { if (w->new) continue; @@ -3394,7 +3396,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return change; } @@ -3499,7 +3501,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_unlock(&card->dapm_mutex); if (ret > 0) - soc_dpcm_runtime_update(card); + snd_soc_dpcm_runtime_update(card); return change; } @@ -3604,6 +3606,9 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, ret = PTR_ERR(w->pinctrl); goto request_failed; } + + /* set to sleep_state when initializing */ + dapm_pinctrl_event(w, NULL, SND_SOC_DAPM_POST_PMD); break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); @@ -3698,6 +3703,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->dapm = dapm; INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); + /* see for_each_card_widgets */ list_add_tail(&w->list, &dapm->card->widgets); snd_soc_dapm_for_each_direction(dir) { @@ -4222,7 +4228,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) struct snd_soc_dai *dai; /* For each DAI widget... */ - list_for_each_entry(dai_w, &card->widgets, list) { + for_each_card_widgets(card, dai_w) { switch (dai_w->id) { case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: @@ -4241,7 +4247,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ - list_for_each_entry(w, &card->widgets, list) { + for_each_card_widgets(card, w) { if (w->dapm != dai_w->dapm) continue; @@ -4271,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) +static void dapm_add_valid_dai_widget(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - int i; if (rtd->dai_link->params) { playback_cpu = cpu_dai->capture_widget; @@ -4292,77 +4297,92 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture_cpu = capture; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI playback if widgets are valid */ - codec = codec_dai->playback_widget; - - if (playback_cpu && codec) { - if (!playback) { - substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(playback)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + /* connect BE DAI playback if widgets are valid */ + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + playback = snd_soc_dapm_new_dai(card, substream, + "playback"); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + goto capture; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); } - } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI capture if widgets are valid */ - codec = codec_dai->capture_widget; - - if (codec && capture_cpu) { - if (!capture) { - substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(capture)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); + + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); + } + +capture: + /* connect BE DAI capture if widgets are valid */ + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; + capture = snd_soc_dapm_new_dai(card, substream, + "capture"); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + return; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); } + + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); + + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai; + int i; + + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dais(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } + +} + static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { struct snd_soc_dapm_widget *w; unsigned int ep; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + w = snd_soc_dai_get_widget(dai, stream); if (w) { dapm_mark_dirty(w, "stream event"); @@ -4413,12 +4433,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i; - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); - for_each_rtd_codec_dai(rtd, i, codec_dai) - soc_dapm_dai_stream_event(codec_dai, stream, event); + for_each_rtd_dais(rtd, i, dai) + soc_dapm_dai_stream_event(dai, stream, event); dapm_power_widgets(rtd->card, event); } @@ -4754,6 +4773,7 @@ void snd_soc_dapm_init(struct snd_soc_dapm_context *dapm, } INIT_LIST_HEAD(&dapm->list); + /* see for_each_card_dapms */ list_add(&dapm->list, &card->dapm_list); } EXPORT_SYMBOL_GPL(snd_soc_dapm_init); @@ -4767,7 +4787,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) mutex_lock(&card->dapm_mutex); - list_for_each_entry(w, &dapm->card->widgets, list) { + for_each_card_widgets(dapm->card, w) { if (w->dapm != dapm) continue; if (w->power) { @@ -4800,7 +4820,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { struct snd_soc_dapm_context *dapm; - list_for_each_entry(dapm, &card->dapm_list, list) { + for_each_card_dapms(card, dapm) { if (dapm != &card->dapm) { soc_dapm_shutdown_dapm(dapm); if (dapm->bias_level == SND_SOC_BIAS_STANDBY) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 2cc25651661c..facf1922a714 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -62,6 +62,12 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_dmaengine_dai_dma_data *dma_data; int ret; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); @@ -118,6 +124,12 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_hardware hw; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -185,6 +197,12 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_dmaengine_dai_dma_data *dma_data; dma_filter_fn fn = NULL; + if (rtd->num_cpus > 1) { + dev_err(rtd->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return NULL; + } + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) @@ -237,7 +255,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, max_buffer_size = SIZE_MAX; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + for_each_pcm_streams(i) { substream = rtd->pcm->streams[i].substream; if (!substream) continue; @@ -371,8 +389,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, dev = config->dma_dev; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) name = "rx-tx"; else @@ -401,8 +418,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) { unsigned int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (!pcm->chan[i]) continue; dma_release_channel(pcm->chan[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..e256d438ee68 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -28,6 +28,180 @@ #define DPCM_MAX_BE_USERS 8 +#ifdef CONFIG_DEBUG_FS +static const char *dpcm_state_string(enum snd_soc_dpcm_state state) +{ + switch (state) { + case SND_SOC_DPCM_STATE_NEW: + return "new"; + case SND_SOC_DPCM_STATE_OPEN: + return "open"; + case SND_SOC_DPCM_STATE_HW_PARAMS: + return "hw_params"; + case SND_SOC_DPCM_STATE_PREPARE: + return "prepare"; + case SND_SOC_DPCM_STATE_START: + return "start"; + case SND_SOC_DPCM_STATE_STOP: + return "stop"; + case SND_SOC_DPCM_STATE_SUSPEND: + return "suspend"; + case SND_SOC_DPCM_STATE_PAUSED: + return "paused"; + case SND_SOC_DPCM_STATE_HW_FREE: + return "hw_free"; + case SND_SOC_DPCM_STATE_CLOSE: + return "close"; + } + + return "unknown"; +} + +static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, + int stream, char *buf, size_t size) +{ + struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; + struct snd_soc_dpcm *dpcm; + ssize_t offset = 0; + unsigned long flags; + + /* FE state */ + offset += scnprintf(buf + offset, size - offset, + "[%s - %s]\n", fe->dai_link->name, + stream ? "Capture" : "Playback"); + + offset += scnprintf(buf + offset, size - offset, "State: %s\n", + dpcm_state_string(fe->dpcm[stream].state)); + + if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += scnprintf(buf + offset, size - offset, + "Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + + /* BEs state */ + offset += scnprintf(buf + offset, size - offset, "Backends:\n"); + + if (list_empty(&fe->dpcm[stream].be_clients)) { + offset += scnprintf(buf + offset, size - offset, + " No active DSP links\n"); + goto out; + } + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + for_each_dpcm_be(fe, stream, dpcm) { + struct snd_soc_pcm_runtime *be = dpcm->be; + params = &dpcm->hw_params; + + offset += scnprintf(buf + offset, size - offset, + "- %s\n", be->dai_link->name); + + offset += scnprintf(buf + offset, size - offset, + " State: %s\n", + dpcm_state_string(be->dpcm[stream].state)); + + if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && + (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) + offset += scnprintf(buf + offset, size - offset, + " Hardware Params: " + "Format = %s, Channels = %d, Rate = %d\n", + snd_pcm_format_name(params_format(params)), + params_channels(params), + params_rate(params)); + } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); +out: + return offset; +} + +static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_pcm_runtime *fe = file->private_data; + ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + int stream; + char *buf; + + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + + buf = kmalloc(out_count, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + for_each_pcm_streams(stream) + if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + offset += dpcm_show_state(fe, stream, + buf + offset, + out_count - offset); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); + + kfree(buf); + return ret; +} + +static const struct file_operations dpcm_state_fops = { + .open = simple_open, + .read = dpcm_state_read_file, + .llseek = default_llseek, +}; + +void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd->dai_link) + return; + + if (!rtd->dai_link->dynamic) + return; + + if (!rtd->card->debugfs_card_root) + return; + + rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, + rtd->card->debugfs_card_root); + + debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); +} + +static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream) +{ + char *name; + + name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir( + name, dpcm->fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } +} + +static void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) +{ + debugfs_remove_recursive(dpcm->debugfs_state); +} + +#else +static inline void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, + int stream) +{ +} + +static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) +{ +} +#endif + static int soc_rtd_startup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { @@ -82,6 +256,21 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, return 0; } +static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, + int stream, int action) +{ + struct snd_soc_dai *dai; + int i; + + lockdep_assert_held(&rtd->card->pcm_mutex); + + for_each_rtd_dais(rtd, i, dai) { + dai->stream_active[stream] += action; + dai->active += action; + dai->component->active += action; + } +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -94,29 +283,9 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, */ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int i; - - lockdep_assert_held(&rtd->card->pcm_mutex); - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active++; - } - - cpu_dai->active++; - cpu_dai->component->active++; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - codec_dai->active++; - codec_dai->component->active++; - } + snd_soc_runtime_action(rtd, stream, 1); } +EXPORT_SYMBOL_GPL(snd_soc_runtime_activate); /** * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components @@ -130,29 +299,9 @@ void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) */ void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int i; - - lockdep_assert_held(&rtd->card->pcm_mutex); - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active--; - } - - cpu_dai->active--; - cpu_dai->component->active--; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - codec_dai->component->active--; - codec_dai->active--; - } + snd_soc_runtime_action(rtd, stream, -1); } +EXPORT_SYMBOL_GPL(snd_soc_runtime_deactivate); /** * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay @@ -287,8 +436,8 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; + struct snd_soc_dai *cpu_dai; unsigned int rate, channels, sample_bits, symmetry, i; rate = params_rate(params); @@ -296,40 +445,51 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, sample_bits = snd_pcm_format_physical_width(params_format(params)); /* reject unmatched parameters when applying symmetry */ - symmetry = cpu_dai->driver->symmetric_rates || - rtd->dai_link->symmetric_rates; + symmetry = rtd->dai_link->symmetric_rates; - for_each_rtd_codec_dai(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_rates; + for_each_rtd_cpu_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_rates; - if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { - dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", - cpu_dai->rate, rate); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (cpu_dai->rate && cpu_dai->rate != rate) { + dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", + cpu_dai->rate, rate); + return -EINVAL; + } + } } - symmetry = cpu_dai->driver->symmetric_channels || - rtd->dai_link->symmetric_channels; + symmetry = rtd->dai_link->symmetric_channels; - for_each_rtd_codec_dai(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_channels; + for_each_rtd_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_channels; - if (symmetry && cpu_dai->channels && cpu_dai->channels != channels) { - dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", - cpu_dai->channels, channels); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (cpu_dai->channels && + cpu_dai->channels != channels) { + dev_err(rtd->dev, "ASoC: unmatched channel symmetry: %d - %d\n", + cpu_dai->channels, channels); + return -EINVAL; + } + } } - symmetry = cpu_dai->driver->symmetric_samplebits || - rtd->dai_link->symmetric_samplebits; + symmetry = rtd->dai_link->symmetric_samplebits; - for_each_rtd_codec_dai(rtd, i, codec_dai) - symmetry |= codec_dai->driver->symmetric_samplebits; + for_each_rtd_dais(rtd, i, dai) + symmetry |= dai->driver->symmetric_samplebits; - if (symmetry && cpu_dai->sample_bits && cpu_dai->sample_bits != sample_bits) { - dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", - cpu_dai->sample_bits, sample_bits); - return -EINVAL; + if (symmetry) { + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (cpu_dai->sample_bits && + cpu_dai->sample_bits != sample_bits) { + dev_err(rtd->dev, "ASoC: unmatched sample bits symmetry: %d - %d\n", + cpu_dai->sample_bits, sample_bits); + return -EINVAL; + } + } } return 0; @@ -338,20 +498,19 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; struct snd_soc_dai_link *link = rtd->dai_link; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; unsigned int symmetry, i; - symmetry = cpu_driver->symmetric_rates || link->symmetric_rates || - cpu_driver->symmetric_channels || link->symmetric_channels || - cpu_driver->symmetric_samplebits || link->symmetric_samplebits; + symmetry = link->symmetric_rates || + link->symmetric_channels || + link->symmetric_samplebits; - for_each_rtd_codec_dai(rtd, i, codec_dai) + for_each_rtd_dais(rtd, i, dai) symmetry = symmetry || - codec_dai->driver->symmetric_rates || - codec_dai->driver->symmetric_channels || - codec_dai->driver->symmetric_samplebits; + dai->driver->symmetric_rates || + dai->driver->symmetric_channels || + dai->driver->symmetric_samplebits; return symmetry; } @@ -373,77 +532,98 @@ static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) static void soc_pcm_apply_msb(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; + struct snd_soc_pcm_stream *pcm_codec, *pcm_cpu; + int stream = substream->stream; int i; - unsigned int bits = 0, cpu_bits; + unsigned int bits = 0, cpu_bits = 0; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->playback.sig_bits == 0) { - bits = 0; - break; - } - bits = max(codec_dai->driver->playback.sig_bits, bits); + for_each_rtd_codec_dais(rtd, i, codec_dai) { + pcm_codec = snd_soc_dai_get_pcm_stream(codec_dai, stream); + + if (pcm_codec->sig_bits == 0) { + bits = 0; + break; } - cpu_bits = cpu_dai->driver->playback.sig_bits; - } else { - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->capture.sig_bits == 0) { - bits = 0; - break; - } - bits = max(codec_dai->driver->capture.sig_bits, bits); + bits = max(pcm_codec->sig_bits, bits); + } + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + pcm_cpu = snd_soc_dai_get_pcm_stream(cpu_dai, stream); + + if (pcm_cpu->sig_bits == 0) { + cpu_bits = 0; + break; } - cpu_bits = cpu_dai->driver->capture.sig_bits; + cpu_bits = max(pcm_cpu->sig_bits, cpu_bits); } soc_pcm_set_msb(substream, bits); soc_pcm_set_msb(substream, cpu_bits); } -static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) +/** + * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream + * @rtd: ASoC PCM runtime + * @hw: PCM hardware parameters (output) + * @stream: Direction of the PCM stream + * + * Calculates the subset of stream parameters supported by all DAIs + * associated with the PCM stream. + */ +int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hardware *hw, int stream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hardware *hw = &runtime->hw; - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai; - struct snd_soc_dai_driver *cpu_dai_drv = rtd->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_dai *cpu_dai; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; unsigned int chan_min = 0, chan_max = UINT_MAX; + unsigned int cpu_chan_min = 0, cpu_chan_max = UINT_MAX; unsigned int rate_min = 0, rate_max = UINT_MAX; - unsigned int rates = UINT_MAX; + unsigned int cpu_rate_min = 0, cpu_rate_max = UINT_MAX; + unsigned int rates = UINT_MAX, cpu_rates = UINT_MAX; u64 formats = ULLONG_MAX; int i; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; + /* first calculate min/max only for CPUs in the DAI link */ + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - /* first calculate min/max only for CODECs in the DAI link */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { + /* + * Skip CPUs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + */ + if (!snd_soc_dai_stream_valid(cpu_dai, stream)) + continue; + + cpu_stream = snd_soc_dai_get_pcm_stream(cpu_dai, stream); + + cpu_chan_min = max(cpu_chan_min, cpu_stream->channels_min); + cpu_chan_max = min(cpu_chan_max, cpu_stream->channels_max); + cpu_rate_min = max(cpu_rate_min, cpu_stream->rate_min); + cpu_rate_max = min_not_zero(cpu_rate_max, cpu_stream->rate_max); + formats &= cpu_stream->formats; + cpu_rates = snd_pcm_rate_mask_intersect(cpu_stream->rates, + cpu_rates); + } + + /* second calculate min/max only for CODECs in the DAI link */ + for_each_rtd_codec_dais(rtd, i, codec_dai) { /* * Skip CODECs which don't support the current stream type. * Otherwise, since the rate, channel, and format values will * zero in that case, we would have no usable settings left, * causing the resulting setup to fail. - * At least one CODEC should match, otherwise we should have - * bailed out on a higher level, since there would be no - * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(codec_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; - codec_dai_drv = codec_dai->driver; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); + chan_min = max(chan_min, codec_stream->channels_min); chan_max = min(chan_max, codec_stream->channels_max); rate_min = max(rate_min, codec_stream->rate_min); @@ -452,74 +632,107 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) rates = snd_pcm_rate_mask_intersect(codec_stream->rates, rates); } + /* Verify both a valid CPU DAI and a valid CODEC DAI were found */ + if (!chan_min || !cpu_chan_min) + return -EINVAL; + /* * chan min/max cannot be enforced if there are multiple CODEC DAIs - * connected to a single CPU DAI, use CPU DAI's directly and let + * connected to CPU DAI(s), use CPU DAI's directly and let * channel allocation be fixed up later */ if (rtd->num_codecs > 1) { - chan_min = cpu_stream->channels_min; - chan_max = cpu_stream->channels_max; + chan_min = cpu_chan_min; + chan_max = cpu_chan_max; } - hw->channels_min = max(chan_min, cpu_stream->channels_min); - hw->channels_max = min(chan_max, cpu_stream->channels_max); - if (hw->formats) - hw->formats &= formats & cpu_stream->formats; - else - hw->formats = formats & cpu_stream->formats; - hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_stream->rates); + /* finally find a intersection between CODECs and CPUs */ + hw->channels_min = max(chan_min, cpu_chan_min); + hw->channels_max = min(chan_max, cpu_chan_max); + hw->formats = formats; + hw->rates = snd_pcm_rate_mask_intersect(rates, cpu_rates); - snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_limit_rates(hw); - hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); + hw->rate_min = max(hw->rate_min, cpu_rate_min); hw->rate_min = max(hw->rate_min, rate_min); - hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, cpu_rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); + + return 0; } +EXPORT_SYMBOL_GPL(snd_soc_runtime_calc_hw); -static int soc_pcm_components_open(struct snd_pcm_substream *substream, - struct snd_soc_component **last) +static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) { + struct snd_pcm_hardware *hw = &substream->runtime->hw; struct snd_soc_pcm_runtime *rtd = substream->private_data; + u64 formats = hw->formats; + + /* + * At least one CPU and one CODEC should match. Otherwise, we should + * have bailed out on a higher level, since there would be no CPU or + * CODEC to support the transfer direction in that case. + */ + snd_soc_runtime_calc_hw(rtd, hw, substream->stream); + + if (formats) + hw->formats &= formats; +} + +static int soc_pcm_components_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *last = NULL; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { - *last = component; + last = component; ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, "ASoC: can't get module %s\n", component->name); - return ret; + break; } ret = snd_soc_component_open(component, substream); if (ret < 0) { + snd_soc_component_module_put_when_close(component); dev_err(component->dev, "ASoC: can't open component %s: %d\n", component->name, ret); - return ret; + break; } } - *last = NULL; - return 0; + + if (ret < 0) { + /* rollback on error */ + for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + + snd_soc_component_close(component, substream); + snd_soc_component_module_put_when_close(component); + } + } + + return ret; } -static int soc_pcm_components_close(struct snd_pcm_substream *substream, - struct snd_soc_component *last) +static int soc_pcm_components_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - int i, ret = 0; + int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { - if (component == last) - break; + r = snd_soc_component_close(component, substream); + if (r < 0) + ret = r; /* use last ret */ - ret |= snd_soc_component_close(component, substream); snd_soc_component_module_put_when_close(component); } @@ -527,6 +740,45 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, } /* + * Called by ALSA when a PCM substream is closed. Private data can be + * freed here. The cpu DAI, codec DAI, machine and components are also + * shutdown. + */ +static int soc_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_dai *dai; + int i; + + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + snd_soc_runtime_deactivate(rtd, substream->stream); + + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_shutdown(dai, substream); + + soc_rtd_shutdown(rtd, substream); + + soc_pcm_components_close(substream); + + snd_soc_dapm_stream_stop(rtd, substream->stream); + + mutex_unlock(&rtd->card->pcm_mutex); + + for_each_rtd_components(rtd, i, component) { + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + } + + for_each_rtd_components(rtd, i, component) + if (!component->active) + pinctrl_pm_select_sleep_state(component->dev); + + return 0; +} + +/* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, component, machine and codec DAI. @@ -536,9 +788,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; const char *codec_dai_name = "multicodec"; + const char *cpu_dai_name = "multicpu"; int i, ret = 0; for_each_rtd_components(rtd, i, component) @@ -549,38 +801,31 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - /* startup the audio subsystem */ - ret = snd_soc_dai_startup(cpu_dai, substream); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", - cpu_dai->name, ret); - goto out; - } - - ret = soc_pcm_components_open(substream, &component); + ret = soc_pcm_components_open(substream); if (ret < 0) goto component_err; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_startup(codec_dai, substream); + ret = soc_rtd_startup(rtd, substream); + if (ret < 0) { + pr_err("ASoC: %s startup failed: %d\n", + rtd->dai_link->name, ret); + goto rtd_startup_err; + } + + /* startup the audio subsystem */ + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_startup(dai, substream); if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't open codec %s: %d\n", - codec_dai->name, ret); - goto codec_dai_err; + dev_err(dai->dev, + "ASoC: can't open DAI %s: %d\n", + dai->name, ret); + goto config_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_dai->tx_mask = 0; + dai->tx_mask = 0; else - codec_dai->rx_mask = 0; - } - - ret = soc_rtd_startup(rtd, substream); - if (ret < 0) { - pr_err("ASoC: %s startup failed: %d\n", - rtd->dai_link->name, ret); - goto machine_err; + dai->rx_mask = 0; } /* Dynamic PCM DAI links compat checks use dynamic capabilities */ @@ -593,46 +838,43 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (rtd->num_codecs == 1) codec_dai_name = rtd->codec_dai->name; + if (rtd->num_cpus == 1) + cpu_dai_name = rtd->cpu_dai->name; + if (soc_pcm_has_symmetry(substream)) runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "ASoC: %s <-> %s No matching formats\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max || runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "ASoC: %s <-> %s No matching channels\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); goto config_err; } soc_pcm_apply_msb(substream); /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active) { - ret = soc_pcm_apply_symmetry(substream, cpu_dai); - if (ret != 0) - goto config_err; - } - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream, codec_dai); + for_each_rtd_dais(rtd, i, dai) { + if (dai->active) { + ret = soc_pcm_apply_symmetry(substream, dai); if (ret != 0) goto config_err; } } pr_debug("ASoC: %s <-> %s info:\n", - codec_dai_name, cpu_dai->name); + codec_dai_name, cpu_dai_name); pr_debug("ASoC: rate mask 0x%x\n", runtime->hw.rates); pr_debug("ASoC: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -647,20 +889,13 @@ dynamic: return 0; config_err: - soc_rtd_shutdown(rtd, substream); - -machine_err: - i = rtd->num_codecs; - -codec_dai_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) - snd_soc_dai_shutdown(codec_dai, substream); + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_shutdown(dai, substream); + soc_rtd_shutdown(rtd, substream); +rtd_startup_err: + soc_pcm_components_close(substream); component_err: - soc_pcm_components_close(substream, component); - - snd_soc_dai_shutdown(cpu_dai, substream); -out: mutex_unlock(&rtd->card->pcm_mutex); for_each_rtd_components(rtd, i, component) { @@ -686,59 +921,6 @@ static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) } /* - * Called by ALSA when a PCM substream is closed. Private data can be - * freed here. The cpu DAI, codec DAI, machine and components are also - * shutdown. - */ -static int soc_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int i; - - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - - snd_soc_runtime_deactivate(rtd, substream->stream); - - /* clear the corresponding DAIs rate when inactive */ - if (!cpu_dai->active) - cpu_dai->rate = 0; - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (!codec_dai->active) - codec_dai->rate = 0; - } - - snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); - - snd_soc_dai_shutdown(cpu_dai, substream); - - for_each_rtd_codec_dai(rtd, i, codec_dai) - snd_soc_dai_shutdown(codec_dai, substream); - - soc_rtd_shutdown(rtd, substream); - - soc_pcm_components_close(substream, NULL); - - snd_soc_dapm_stream_stop(rtd, substream->stream); - - mutex_unlock(&rtd->card->pcm_mutex); - - for_each_rtd_components(rtd, i, component) { - pm_runtime_mark_last_busy(component->dev); - pm_runtime_put_autosuspend(component->dev); - } - - for_each_rtd_components(rtd, i, component) - if (!component->active) - pinctrl_pm_select_sleep_state(component->dev); - - return 0; -} - -/* * Called by ALSA when the PCM substream is prepared, can set format, sample * rate, etc. This function is non atomic and can be called multiple times, * it can refer to the runtime info. @@ -747,8 +929,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -769,23 +950,15 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_prepare(codec_dai, substream); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_prepare(dai, substream); if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: codec DAI prepare error: %d\n", - ret); + dev_err(dai->dev, + "ASoC: DAI prepare error: %d\n", ret); goto out; } } - ret = snd_soc_dai_prepare(cpu_dai, substream); - if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: cpu DAI prepare error: %d\n", ret); - goto out; - } - /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && rtd->pop_wait) { @@ -796,10 +969,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(rtd, substream->stream, SND_SOC_DAPM_STREAM_START); - for_each_rtd_codec_dai(rtd, i, codec_dai) - snd_soc_dai_digital_mute(codec_dai, 0, - substream->stream); - snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); + for_each_rtd_dais(rtd, i, dai) + snd_soc_dai_digital_mute(dai, 0, substream->stream); out: mutex_unlock(&rtd->card->pcm_mutex); @@ -822,13 +993,15 @@ static int soc_pcm_components_hw_free(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - int i, ret = 0; + int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { if (component == last) break; - ret |= snd_soc_component_hw_free(component, substream); + r = snd_soc_component_hw_free(component, substream); + if (r < 0) + ret = r; /* use last ret */ } return ret; @@ -844,7 +1017,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret = 0; @@ -861,7 +1034,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, goto out; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { struct snd_pcm_hw_params codec_params; /* @@ -908,17 +1081,26 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_dapm_update_dai(substream, &codec_params, codec_dai); } - ret = snd_soc_dai_hw_params(cpu_dai, substream, params); - if (ret < 0) - goto interface_err; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; - /* store the parameters for each DAIs */ - cpu_dai->rate = params_rate(params); - cpu_dai->channels = params_channels(params); - cpu_dai->sample_bits = - snd_pcm_format_physical_width(params_format(params)); + ret = snd_soc_dai_hw_params(cpu_dai, substream, params); + if (ret < 0) + goto interface_err; + + /* store the parameters for each DAI */ + cpu_dai->rate = params_rate(params); + cpu_dai->channels = params_channels(params); + cpu_dai->sample_bits = + snd_pcm_format_physical_width(params_format(params)); - snd_soc_dapm_update_dai(substream, params, cpu_dai); + snd_soc_dapm_update_dai(substream, params, cpu_dai); + } for_each_rtd_components(rtd, i, component) { ret = snd_soc_component_hw_params(component, substream, params); @@ -938,14 +1120,21 @@ out: component_err: soc_pcm_components_hw_free(substream, component); - snd_soc_dai_hw_free(cpu_dai, substream); - cpu_dai->rate = 0; + i = rtd->num_cpus; interface_err: + for_each_rtd_cpu_dais_rollback(rtd, i, cpu_dai) { + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + + snd_soc_dai_hw_free(cpu_dai, substream); + cpu_dai->rate = 0; + } + i = rtd->num_codecs; codec_err: - for_each_rtd_codec_dai_rollback(rtd, i, codec_dai) { + for_each_rtd_codec_dais_rollback(rtd, i, codec_dai) { if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) continue; @@ -965,34 +1154,23 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_dai *dai; int i; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); /* clear the corresponding DAIs parameters when going to be inactive */ - if (cpu_dai->active == 1) { - cpu_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; - } + for_each_rtd_dais(rtd, i, dai) { + int active = dai->stream_active[substream->stream]; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->active == 1) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; + if (dai->active == 1) { + dai->rate = 0; + dai->channels = 0; + dai->sample_bits = 0; } - } - /* apply codec digital mute */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) - snd_soc_dai_digital_mute(codec_dai, 1, - substream->stream); + if (active == 1) + snd_soc_dai_digital_mute(dai, 1, substream->stream); } /* free any machine hw params */ @@ -1002,15 +1180,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) soc_pcm_components_hw_free(substream, NULL); /* now free hw params for the DAIs */ - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + for_each_rtd_dais(rtd, i, dai) { + if (!snd_soc_dai_stream_valid(dai, substream->stream)) continue; - snd_soc_dai_hw_free(codec_dai, substream); + snd_soc_dai_hw_free(dai, substream); } - snd_soc_dai_hw_free(cpu_dai, substream); - mutex_unlock(&rtd->card->pcm_mutex); return 0; } @@ -1019,8 +1195,7 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; ret = soc_rtd_trigger(rtd, substream, cmd); @@ -1033,12 +1208,8 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) return ret; } - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_trigger(dai, substream, cmd); if (ret < 0) return ret; } @@ -1050,20 +1221,15 @@ static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_trigger(dai, substream, cmd); if (ret < 0) return ret; } - ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; - for_each_rtd_components(rtd, i, component) { ret = snd_soc_component_trigger(component, substream, cmd); if (ret < 0) @@ -1103,20 +1269,15 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_bespoke_trigger(codec_dai, substream, cmd); + for_each_rtd_dais(rtd, i, dai) { + ret = snd_soc_dai_bespoke_trigger(dai, substream, cmd); if (ret < 0) return ret; } - ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); - if (ret < 0) - return ret; - return 0; } /* @@ -1127,12 +1288,13 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; snd_pcm_sframes_t codec_delay = 0; + snd_pcm_sframes_t cpu_delay = 0; int i; /* clearing the previous total delay */ @@ -1143,9 +1305,13 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) /* base delay if assigned in pointer callback */ delay = runtime->delay; - delay += snd_soc_dai_delay(cpu_dai, substream); + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + cpu_delay = max(cpu_delay, + snd_soc_dai_delay(cpu_dai, substream)); + } + delay += cpu_delay; - for_each_rtd_codec_dai(rtd, i, codec_dai) { + for_each_rtd_codec_dais(rtd, i, codec_dai) { codec_delay = max(codec_delay, snd_soc_dai_delay(codec_dai, substream)); } @@ -1162,9 +1328,6 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; -#ifdef CONFIG_DEBUG_FS - char *name; -#endif /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1189,17 +1352,8 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); -#ifdef CONFIG_DEBUG_FS - name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, - stream ? "capture" : "playback"); - if (name) { - dpcm->debugfs_state = debugfs_create_dir(name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, - &dpcm->state); - kfree(name); - } -#endif + dpcm_create_debugfs_state(dpcm, stream); + return 1; } @@ -1252,9 +1406,8 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) /* BEs still alive need new FE */ dpcm_be_reparent(fe, dpcm->be, stream); -#ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(dpcm->debugfs_state); -#endif + dpcm_remove_debugfs_state(dpcm); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); @@ -1268,74 +1421,41 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; + struct snd_soc_dapm_widget *w; struct snd_soc_dai *dai; int i; dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - for_each_card_rtds(card, be) { - - if (!be->dai_link->no_pcm) - continue; - - dev_dbg(card->dev, "ASoC: try BE : %s\n", - be->cpu_dai->playback_widget ? - be->cpu_dai->playback_widget->name : "(not set)"); + for_each_card_rtds(card, be) { - if (be->cpu_dai->playback_widget == widget) - return be; - - for_each_rtd_codec_dai(be, i, dai) { - if (dai->playback_widget == widget) - return be; - } - } - } else { - - for_each_card_rtds(card, be) { + if (!be->dai_link->no_pcm) + continue; - if (!be->dai_link->no_pcm) - continue; + for_each_rtd_dais(be, i, dai) { + w = snd_soc_dai_get_widget(dai, stream); - dev_dbg(card->dev, "ASoC: try BE %s\n", - be->cpu_dai->capture_widget ? - be->cpu_dai->capture_widget->name : "(not set)"); + dev_dbg(card->dev, "ASoC: try BE : %s\n", + w ? w->name : "(not set)"); - if (be->cpu_dai->capture_widget == widget) + if (w == widget) return be; - - for_each_rtd_codec_dai(be, i, dai) { - if (dai->capture_widget == widget) - return be; - } } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } -static inline struct snd_soc_dapm_widget * - dai_get_widget(struct snd_soc_dai *dai, int stream) -{ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - return dai->playback_widget; - else - return dai->capture_widget; -} - static int widget_in_list(struct snd_soc_dapm_widget_list *list, struct snd_soc_dapm_widget *widget) { + struct snd_soc_dapm_widget *w; int i; - for (i = 0; i < list->num_widgets; i++) { - if (widget == list->widgets[i]) + for_each_dapm_widgets(list, i, w) + if (widget == w) return 1; - } return 0; } @@ -1345,36 +1465,17 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *dai; - int i; + int stream; - if (dir == SND_SOC_DAPM_DIR_OUT) { - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; - - if (rtd->cpu_dai->playback_widget == widget) - return true; - - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_widget == widget) - return true; - } - } - } else { /* SND_SOC_DAPM_DIR_IN */ - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; - - if (rtd->cpu_dai->capture_widget == widget) - return true; + /* adjust dir to stream */ + if (dir == SND_SOC_DAPM_DIR_OUT) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->capture_widget == widget) - return true; - } - } - } + rtd = dpcm_get_be(card, widget, stream); + if (rtd) + return true; return false; } @@ -1385,6 +1486,12 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, struct snd_soc_dai *cpu_dai = fe->cpu_dai; int paths; + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } + /* get number of valid DAI paths and their widgets */ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, dpcm_end_walk_at_be); @@ -1395,37 +1502,42 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, - struct snd_soc_dapm_widget_list **list_) +void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + snd_soc_dapm_dai_free_widgets(list); +} + +static bool dpcm_be_is_active(struct snd_soc_dpcm *dpcm, int stream, + struct snd_soc_dapm_widget_list *list) { - struct snd_soc_dpcm *dpcm; - struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_dapm_widget *widget; struct snd_soc_dai *dai; - int prune = 0; - int do_prune; - - /* Destroy any old FE <--> BE connections */ - for_each_dpcm_be(fe, stream, dpcm) { - unsigned int i; + unsigned int i; - /* is there a valid CPU DAI widget for this BE */ - widget = dai_get_widget(dpcm->be->cpu_dai, stream); + /* is there a valid DAI widget for this BE */ + for_each_rtd_dais(dpcm->be, i, dai) { + widget = snd_soc_dai_get_widget(dai, stream); - /* prune the BE if it's no longer in our active list */ + /* + * The BE is pruned only if none of the dai + * widgets are in the active list. + */ if (widget && widget_in_list(list, widget)) - continue; + return true; + } - /* is there a valid CODEC DAI widget for this BE */ - do_prune = 1; - for_each_rtd_codec_dai(dpcm->be, i, dai) { - widget = dai_get_widget(dai, stream); + return false; +} - /* prune the BE if it's no longer in our active list */ - if (widget && widget_in_list(list, widget)) - do_prune = 0; - } - if (!do_prune) +static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, + struct snd_soc_dapm_widget_list **list_) +{ + struct snd_soc_dpcm *dpcm; + int prune = 0; + + /* Destroy any old FE <--> BE connections */ + for_each_dpcm_be(fe, stream, dpcm) { + if (dpcm_be_is_active(dpcm, stream, *list_)) continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", @@ -1446,12 +1558,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_card *card = fe->card; struct snd_soc_dapm_widget_list *list = *list_; struct snd_soc_pcm_runtime *be; + struct snd_soc_dapm_widget *widget; int i, new = 0, err; /* Create any new FE <--> BE connections */ - for (i = 0; i < list->num_widgets; i++) { + for_each_dapm_widgets(list, i, widget) { - switch (list->widgets[i]->id) { + switch (widget->id) { case snd_soc_dapm_dai_in: if (stream != SNDRV_PCM_STREAM_PLAYBACK) continue; @@ -1465,17 +1578,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, } /* is there a valid BE rtd for this widget */ - be = dpcm_get_be(card, list->widgets[i], stream); + be = dpcm_get_be(card, widget, stream); if (!be) { dev_err(fe->dev, "ASoC: no BE found for %s\n", - list->widgets[i]->name); + widget->name); continue; } - /* make sure BE is a real BE */ - if (!be->dai_link->no_pcm) - continue; - /* don't connect if FE is not running */ if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; @@ -1484,7 +1593,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, err = dpcm_be_connect(fe, be, stream); if (err < 0) { dev_err(fe->dev, "ASoC: can't connect %s\n", - list->widgets[i]->name); + widget->name); break; } else if (err == 0) /* already connected */ continue; @@ -1671,11 +1780,10 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; int i; - for_each_rtd_codec_dai(be, i, dai) { + for_each_rtd_codec_dais(be, i, dai) { /* * Skip CODECs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details @@ -1683,11 +1791,7 @@ static void dpcm_runtime_merge_format(struct snd_pcm_substream *substream, if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = dai->driver; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(dai, stream); *formats &= codec_stream->formats; } @@ -1712,30 +1816,33 @@ static void dpcm_runtime_merge_chan(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_dai *dai; + int i; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; + for_each_rtd_cpu_dais(be, i, dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(dai, stream)) + continue; + + cpu_stream = snd_soc_dai_get_pcm_stream(dai, stream); - *channels_min = max(*channels_min, cpu_stream->channels_min); - *channels_max = min(*channels_max, cpu_stream->channels_max); + *channels_min = max(*channels_min, + cpu_stream->channels_min); + *channels_max = min(*channels_max, + cpu_stream->channels_max); + } /* * chan min/max cannot be enforced if there are multiple CODEC * DAIs connected to a single CPU DAI, use CPU DAI's directly */ if (be->num_codecs == 1) { - codec_dai_drv = be->codec_dais[0]->driver; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + codec_stream = snd_soc_dai_get_pcm_stream(be->codec_dais[0], stream); *channels_min = max(*channels_min, codec_stream->channels_min); @@ -1764,41 +1871,23 @@ static void dpcm_runtime_merge_rate(struct snd_pcm_substream *substream, for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; - struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; - struct snd_soc_dai_driver *codec_dai_drv; - struct snd_soc_pcm_stream *codec_stream; - struct snd_soc_pcm_stream *cpu_stream; + struct snd_soc_pcm_stream *pcm; struct snd_soc_dai *dai; int i; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_stream = &cpu_dai_drv->playback; - else - cpu_stream = &cpu_dai_drv->capture; - - *rate_min = max(*rate_min, cpu_stream->rate_min); - *rate_max = min_not_zero(*rate_max, cpu_stream->rate_max); - *rates = snd_pcm_rate_mask_intersect(*rates, cpu_stream->rates); - - for_each_rtd_codec_dai(be, i, dai) { + for_each_rtd_dais(be, i, dai) { /* - * Skip CODECs which don't support the current stream + * Skip DAIs which don't support the current stream * type. See soc_pcm_init_runtime_hw() for more details */ if (!snd_soc_dai_stream_valid(dai, stream)) continue; - codec_dai_drv = dai->driver; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &codec_dai_drv->playback; - else - codec_stream = &codec_dai_drv->capture; + pcm = snd_soc_dai_get_pcm_stream(dai, stream); - *rate_min = max(*rate_min, codec_stream->rate_min); - *rate_max = min_not_zero(*rate_max, - codec_stream->rate_max); - *rates = snd_pcm_rate_mask_intersect(*rates, - codec_stream->rates); + *rate_min = max(*rate_min, pcm->rate_min); + *rate_max = min_not_zero(*rate_max, pcm->rate_max); + *rates = snd_pcm_rate_mask_intersect(*rates, pcm->rates); } } } @@ -1807,13 +1896,21 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai *cpu_dai; + int i; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); - else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + /* + * Skip CPUs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(cpu_dai, substream->stream)) + continue; + + dpcm_init_runtime_hw(runtime, + snd_soc_dai_get_pcm_stream(cpu_dai, + substream->stream)); + } dpcm_runtime_merge_format(substream, &runtime->hw.formats); dpcm_runtime_merge_chan(substream, &runtime->hw.channels_min, @@ -1850,18 +1947,21 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, { struct snd_soc_dpcm *dpcm; struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dai *fe_cpu_dai = fe->cpu_dai; + struct snd_soc_dai *fe_cpu_dai; int err; + int i; /* apply symmetry for FE */ if (soc_pcm_has_symmetry(fe_substream)) fe_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; - /* Symmetry only applies if we've got an active stream. */ - if (fe_cpu_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); - if (err < 0) - return err; + for_each_rtd_cpu_dais (fe, i, fe_cpu_dai) { + /* Symmetry only applies if we've got an active stream. */ + if (fe_cpu_dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, fe_cpu_dai); + if (err < 0) + return err; + } } /* apply symmetry for BE */ @@ -1870,7 +1970,7 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; + struct snd_soc_dai *dai; int i; /* A backend may not have the requested substream */ @@ -1885,17 +1985,9 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, be_substream->runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; /* Symmetry only applies if we've got an active stream. */ - if (rtd->cpu_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, - rtd->cpu_dai); - if (err < 0) - return err; - } - - for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->active) { - err = soc_pcm_apply_symmetry(fe_substream, - codec_dai); + for_each_rtd_dais(rtd, i, dai) { + if (dai->active) { + err = soc_pcm_apply_symmetry(fe_substream, dai); if (err < 0) return err; } @@ -1913,7 +2005,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - ret = dpcm_be_dai_startup(fe, fe_substream->stream); + ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret); goto be_err; @@ -1934,17 +2026,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) snd_pcm_limit_hw_rates(runtime); ret = dpcm_apply_symmetry(fe_substream, stream); - if (ret < 0) { + if (ret < 0) dev_err(fe->dev, "ASoC: failed to apply dpcm symmetry %d\n", ret); - goto unwind; - } - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - return 0; unwind: - dpcm_be_dai_startup_unwind(fe, fe_substream->stream); + if (ret < 0) + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; @@ -1998,7 +2086,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ - dpcm_be_dai_shutdown(fe, substream->stream); + dpcm_be_dai_shutdown(fe, stream); dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); @@ -2176,9 +2264,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - memcpy(&fe->dpcm[substream->stream].hw_params, params, + memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); - ret = dpcm_be_dai_hw_params(fe, substream->stream); + ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; @@ -2500,7 +2588,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - ret = dpcm_be_dai_prepare(fe, substream->stream); + ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; @@ -2652,36 +2740,18 @@ disconnect: return ret; } -static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) -{ - int ret; - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); - ret = dpcm_run_update_startup(fe, stream); - if (ret < 0) - dev_err(fe->dev, "ASoC: failed to startup some BEs\n"); - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - - return ret; -} - -static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) -{ - int ret; - - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); - ret = dpcm_run_update_shutdown(fe, stream); - if (ret < 0) - dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); - dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - - return ret; -} - static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { struct snd_soc_dapm_widget_list *list; + int stream; int count, paths; + int ret; + + if (fe->num_cpus > 1) { + dev_err(fe->dev, + "%s doesn't support Multi CPU yet\n", __func__); + return -EINVAL; + } if (!fe->dai_link->dynamic) return 0; @@ -2694,74 +2764,53 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", new ? "new" : "old", fe->dai_link->name); - /* skip if FE doesn't have playback capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - return paths; - } - - /* update any playback paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + for_each_pcm_streams(stream) { - dpcm_path_put(&list); + /* skip if FE doesn't have playback/capture capability */ + if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) || + !snd_soc_dai_stream_valid(fe->codec_dai, stream)) + continue; -capture: - /* skip if FE doesn't have capture capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) - return 0; + /* skip if FE isn't currently playing/capturing */ + if (!fe->cpu_dai->stream_active[stream] || + !fe->codec_dai->stream_active[stream]) + continue; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) - return 0; + paths = dpcm_path_get(fe, stream, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, + stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return paths; + } - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - return paths; - } + /* update any playback/capture paths */ + count = dpcm_process_paths(fe, stream, &list, new); + if (count) { + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); + if (new) + ret = dpcm_run_update_startup(fe, stream); + else + ret = dpcm_run_update_shutdown(fe, stream); + if (ret < 0) + dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - /* update any old capture paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, stream); + dpcm_be_disconnect(fe, stream); + } - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_path_put(&list); } - dpcm_path_put(&list); - return 0; } /* Called by DAPM mixer/mux changes to update audio routing between PCMs and * any DAI links. */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *fe; int ret = 0; @@ -2785,38 +2834,40 @@ out: mutex_unlock(&card->mutex); return ret; } -int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) +EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update); + +static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream) { + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; - struct snd_soc_dai *dai; + int stream = fe_substream->stream; - for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + /* mark FE's links ready to prune */ + for_each_dpcm_be(fe, stream, dpcm) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - struct snd_soc_pcm_runtime *be = dpcm->be; - int i; + dpcm_be_disconnect(fe, stream); - if (be->dai_link->ignore_suspend) - continue; + fe->dpcm[stream].runtime = NULL; +} - for_each_rtd_codec_dai(be, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +{ + struct snd_soc_pcm_runtime *fe = fe_substream->private_data; + int ret; - dev_dbg(be->dev, "ASoC: BE digital mute %s\n", - be->dai_link->name); + mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + ret = dpcm_fe_dai_shutdown(fe_substream); - if (drv->ops && drv->ops->digital_mute && - dai->playback_active) - drv->ops->digital_mute(dai, mute); - } - } + dpcm_fe_dai_cleanup(fe_substream); - return 0; + mutex_unlock(&fe->card->mutex); + return ret; } static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; int ret; int stream = fe_substream->stream; @@ -2826,8 +2877,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) ret = dpcm_path_get(fe, stream, &list); if (ret < 0) { - mutex_unlock(&fe->card->mutex); - return ret; + goto open_end; } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); @@ -2837,37 +2887,12 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_process_paths(fe, stream, &list, 1); ret = dpcm_fe_dai_startup(fe_substream); - if (ret < 0) { - /* clean up all links */ - for_each_dpcm_be(fe, stream, dpcm) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - } + if (ret < 0) + dpcm_fe_dai_cleanup(fe_substream); dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); - mutex_unlock(&fe->card->mutex); - return ret; -} - -static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) -{ - struct snd_soc_pcm_runtime *fe = fe_substream->private_data; - struct snd_soc_dpcm *dpcm; - int stream = fe_substream->stream, ret; - - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - ret = dpcm_fe_dai_shutdown(fe_substream); - - /* mark FE's links ready to prune */ - for_each_dpcm_be(fe, stream, dpcm) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - - fe->dpcm[stream].runtime = NULL; +open_end: mutex_unlock(&fe->card->mutex); return ret; } @@ -2876,7 +2901,7 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai; struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; @@ -2888,22 +2913,29 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (rtd->num_cpus == 1) { + cpu_dai = rtd->cpu_dais[0]; + } else if (rtd->num_cpus == rtd->num_codecs) { + cpu_dai = rtd->cpu_dais[i]; + } else { + dev_err(rtd->card->dev, + "N cpus to M codecs link is not supported yet\n"); + return -EINVAL; + } - for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { @@ -3017,7 +3049,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, - cpu_dai->name); + (rtd->num_cpus > 1) ? "multicpu" : rtd->cpu_dai->name); return ret; } @@ -3050,33 +3082,17 @@ struct snd_pcm_substream * } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); -/* get the BE runtime state */ -enum snd_soc_dpcm_state - snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) -{ - return be->dpcm[stream].state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); - -/* set the BE runtime state */ -void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, - int stream, enum snd_soc_dpcm_state state) -{ - be->dpcm[stream].state = state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); - -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, + int stream, + const enum snd_soc_dpcm_state *states, + int num_states) { struct snd_soc_dpcm *dpcm; int state; int ret = 1; unsigned long flags; + int i; spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { @@ -3085,18 +3101,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, continue; state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) { - ret = 0; - break; + for (i = 0; i < num_states; i++) { + if (state == states[i]) { + ret = 0; + break; + } } } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - /* it's safe to free/stop this BE DAI */ + /* it's safe to do this BE DAI */ return ret; } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -3106,168 +3138,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { - struct snd_soc_dpcm *dpcm; - int state; - int ret = 1; - unsigned long flags; - - spin_lock_irqsave(&fe->card->dpcm_lock, flags); - for_each_dpcm_fe(be, stream, dpcm) { - - if (dpcm->fe == fe) - continue; + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_PREPARE, + }; - state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) { - ret = 0; - break; - } - } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - - /* it's safe to change hw_params */ - return ret; + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); - -#ifdef CONFIG_DEBUG_FS -static const char *dpcm_state_string(enum snd_soc_dpcm_state state) -{ - switch (state) { - case SND_SOC_DPCM_STATE_NEW: - return "new"; - case SND_SOC_DPCM_STATE_OPEN: - return "open"; - case SND_SOC_DPCM_STATE_HW_PARAMS: - return "hw_params"; - case SND_SOC_DPCM_STATE_PREPARE: - return "prepare"; - case SND_SOC_DPCM_STATE_START: - return "start"; - case SND_SOC_DPCM_STATE_STOP: - return "stop"; - case SND_SOC_DPCM_STATE_SUSPEND: - return "suspend"; - case SND_SOC_DPCM_STATE_PAUSED: - return "paused"; - case SND_SOC_DPCM_STATE_HW_FREE: - return "hw_free"; - case SND_SOC_DPCM_STATE_CLOSE: - return "close"; - } - - return "unknown"; -} - -static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, - int stream, char *buf, size_t size) -{ - struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; - struct snd_soc_dpcm *dpcm; - ssize_t offset = 0; - unsigned long flags; - - /* FE state */ - offset += scnprintf(buf + offset, size - offset, - "[%s - %s]\n", fe->dai_link->name, - stream ? "Capture" : "Playback"); - - offset += scnprintf(buf + offset, size - offset, "State: %s\n", - dpcm_state_string(fe->dpcm[stream].state)); - - if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && - (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += scnprintf(buf + offset, size - offset, - "Hardware Params: " - "Format = %s, Channels = %d, Rate = %d\n", - snd_pcm_format_name(params_format(params)), - params_channels(params), - params_rate(params)); - - /* BEs state */ - offset += scnprintf(buf + offset, size - offset, "Backends:\n"); - - if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += scnprintf(buf + offset, size - offset, - " No active DSP links\n"); - goto out; - } - - spin_lock_irqsave(&fe->card->dpcm_lock, flags); - for_each_dpcm_be(fe, stream, dpcm) { - struct snd_soc_pcm_runtime *be = dpcm->be; - params = &dpcm->hw_params; - - offset += scnprintf(buf + offset, size - offset, - "- %s\n", be->dai_link->name); - - offset += scnprintf(buf + offset, size - offset, - " State: %s\n", - dpcm_state_string(be->dpcm[stream].state)); - - if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && - (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += scnprintf(buf + offset, size - offset, - " Hardware Params: " - "Format = %s, Channels = %d, Rate = %d\n", - snd_pcm_format_name(params_format(params)), - params_channels(params), - params_rate(params)); - } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); -out: - return offset; -} - -static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - struct snd_soc_pcm_runtime *fe = file->private_data; - ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; - char *buf; - - buf = kmalloc(out_count, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, - buf + offset, out_count - offset); - - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, - buf + offset, out_count - offset); - - ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); - - kfree(buf); - return ret; -} - -static const struct file_operations dpcm_state_fops = { - .open = simple_open, - .read = dpcm_state_read_file, - .llseek = default_llseek, -}; - -void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) -{ - if (!rtd->dai_link) - return; - - if (!rtd->dai_link->dynamic) - return; - - if (!rtd->card->debugfs_card_root) - return; - - rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name, - rtd->card->debugfs_card_root); - - debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, - rtd, &dpcm_state_fops); -} -#endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..1f81cd2d29cf 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -251,7 +251,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, { int ret = 0; - if (tplg->comp && tplg->ops && tplg->ops->vendor_load) + if (tplg->ops && tplg->ops->vendor_load) ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", @@ -283,7 +283,7 @@ static int soc_tplg_vendor_load(struct soc_tplg *tplg, static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_load) + if (tplg->ops && tplg->ops->widget_load) return tplg->ops->widget_load(tplg->comp, tplg->index, w, tplg_w); @@ -295,7 +295,7 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { - if (tplg->comp && tplg->ops && tplg->ops->widget_ready) + if (tplg->ops && tplg->ops->widget_ready) return tplg->ops->widget_ready(tplg->comp, tplg->index, w, tplg_w); @@ -307,7 +307,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, struct snd_soc_dai_driver *dai_drv, struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - if (tplg->comp && tplg->ops && tplg->ops->dai_load) + if (tplg->ops && tplg->ops->dai_load) return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, pcm, dai); @@ -318,7 +318,7 @@ static int soc_tplg_dai_load(struct soc_tplg *tplg, static int soc_tplg_dai_link_load(struct soc_tplg *tplg, struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { - if (tplg->comp && tplg->ops && tplg->ops->link_load) + if (tplg->ops && tplg->ops->link_load) return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; @@ -327,7 +327,7 @@ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, /* tell the component driver that all firmware has been loaded in this request */ static void soc_tplg_complete(struct soc_tplg *tplg) { - if (tplg->comp && tplg->ops && tplg->ops->complete) + if (tplg->ops && tplg->ops->complete) tplg->ops->complete(tplg->comp); } @@ -684,7 +684,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_bind_event); static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { - if (tplg->comp && tplg->ops && tplg->ops->control_load) + if (tplg->ops && tplg->ops->control_load) return tplg->ops->control_load(tplg->comp, tplg->index, k, hdr); @@ -1174,7 +1174,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, static int soc_tplg_add_route(struct soc_tplg *tplg, struct snd_soc_dapm_route *route) { - if (tplg->comp && tplg->ops && tplg->ops->dapm_route_load) + if (tplg->ops && tplg->ops->dapm_route_load) return tplg->ops->dapm_route_load(tplg->comp, tplg->index, route); @@ -2564,7 +2564,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, } /* pass control to component driver for optional further init */ - if (tplg->comp && tplg->ops && tplg->ops->manifest) + if (tplg->ops && tplg->ops->manifest) ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ @@ -2736,6 +2736,10 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct soc_tplg tplg; int ret; + /* component needs to exist to keep and reference data while parsing */ + if (!comp) + return -EINVAL; + /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); tplg.fw = fw; @@ -2774,7 +2778,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w, *next_w; - list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { + for_each_card_widgets_safe(dapm->card, w, next_w) { /* make sure we are a widget with correct context */ if (w->dobj.type != SND_SOC_DOBJ_WIDGET || w->dapm != dapm) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 827b0ec92522..4dda4b62509f 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -41,6 +41,15 @@ config SND_SOC_SOF_OF required to enable i.MX8 devices. Say Y if you need this option. If unsure select "N". +config SND_SOC_SOF_DEBUG_PROBES + bool "SOF enable data probing" + select SND_SOC_COMPRESS + help + This option enables the data probing feature that can be used to + gather data directly from specific points of the audio pipeline. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_DEVELOPER_SUPPORT bool "SOF developer options support" depends on EXPERT diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 0a8bc72c28a5..8eca2f85c90e 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,6 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c new file mode 100644 index 000000000000..7354dc6a49cf --- /dev/null +++ b/sound/soc/sof/compress.c @@ -0,0 +1,146 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski@intel.com> +// + +#include <sound/soc.h> +#include "compress.h" +#include "ops.h" +#include "probe.h" + +struct snd_compr_ops sof_probe_compressed_ops = { + .copy = sof_probe_compr_copy, +}; +EXPORT_SYMBOL(sof_probe_compressed_ops); + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + ret = snd_sof_probe_compr_assign(sdev, cstream, dai); + if (ret < 0) { + dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret); + return ret; + } + + sdev->extractor_stream_tag = ret; + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_open); + +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + struct sof_probe_point_desc *desc; + size_t num_desc; + int i, ret; + + /* disconnect all probe points */ + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) { + dev_err(dai->dev, "Failed to get probe points: %d\n", ret); + goto exit; + } + + for (i = 0; i < num_desc; i++) + sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1); + kfree(desc); + +exit: + ret = sof_ipc_probe_deinit(sdev); + if (ret < 0) + dev_err(dai->dev, "Failed to deinit probe: %d\n", ret); + + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; + snd_compr_free_pages(cstream); + + return snd_sof_probe_compr_free(sdev, cstream, dai); +} +EXPORT_SYMBOL(sof_probe_compr_free); + +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); + if (ret < 0) + return ret; + + ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai); + if (ret < 0) + return ret; + + ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag, + rtd->dma_bytes); + if (ret < 0) { + dev_err(dai->dev, "Failed to init probe: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_set_params); + +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai); +} +EXPORT_SYMBOL(sof_probe_compr_trigger); + +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); +} +EXPORT_SYMBOL(sof_probe_compr_pointer); + +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_to_user(buf, ptr, count); + } else { + ret = copy_to_user(buf, ptr, n); + ret += copy_to_user(buf + n, rtd->dma_area, count - n); + } + + if (ret) + return count - ret; + return count; +} +EXPORT_SYMBOL(sof_probe_compr_copy); diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h new file mode 100644 index 000000000000..800f163603e1 --- /dev/null +++ b/sound/soc/sof/compress.h @@ -0,0 +1,31 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski@intel.com> + */ + +#ifndef __SOF_COMPRESS_H +#define __SOF_COMPRESS_H + +#include <sound/compress_driver.h> + +extern struct snd_compr_ops sof_probe_compressed_ops; + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai); +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai); +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count); + +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 34cefbaf2d2a..91acfae7935c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -14,6 +14,9 @@ #include <sound/sof.h> #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" +#endif /* see SOF_DBG_ flags */ int sof_core_debug; @@ -286,12 +289,15 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) /* initialize sof device */ sdev->dev = dev; - /* initialize default D0 sub-state */ - sdev->d0_substate = SOF_DSP_D0I0; + /* initialize default DSP power state */ + sdev->dsp_power_state.state = SOF_DSP_PM_D0; sdev->pdata = plat_data; sdev->first_boot = true; sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; +#endif dev_set_drvdata(dev, sdev); /* check all mandatory ops */ diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d2b3b99d3a20..b5c0d6cf72cc 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -17,6 +17,221 @@ #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @buf: String to split into tokens. + * @delim: String containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + */ +static int +strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns) +{ + char *s; + u32 *data, *tmp; + size_t count = 0; + size_t cap = 32; + int ret = 0; + + *tkns = NULL; + *num_tkns = 0; + data = kcalloc(cap, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + while ((s = strsep(buf, delim)) != NULL) { + ret = kstrtouint(s, 0, data + count); + if (ret) + goto exit; + if (++count >= cap) { + cap *= 2; + tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto exit; + } + data = tmp; + } + } + + if (!count) + goto exit; + *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto exit; + } + *num_tkns = count; + +exit: + kfree(data); + return ret; +} + +static int tokenize_input(const char __user *from, size_t count, + loff_t *ppos, u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = ret >= 0 ? -EIO : ret; + goto exit; + } + + buf[count] = '\0'; + ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_read(struct file *file, + char __user *to, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_desc, len = 0; + char *buf; + int i, ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + buf = kzalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) + goto exit; + + for (i = 0; i < num_desc; i++) { + ret = snprintf(buf + len, PAGE_SIZE - len, + "Id: %#010x Purpose: %d Node id: %#x\n", + desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag); + if (ret < 0) + goto free_desc; + len += ret; + } + + ret = simple_read_from_buffer(to, count, ppos, buf, len); +free_desc: + kfree(desc); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_tkns, bytes; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + bytes = sizeof(*tkns) * num_tkns; + if (!num_tkns || (bytes % sizeof(*desc))) { + ret = -EINVAL; + goto exit; + } + + desc = (struct sof_probe_point_desc *)tkns; + ret = sof_ipc_probe_points_add(sdev, + desc, bytes / sizeof(*desc)); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_fops = { + .open = simple_open, + .read = probe_points_read, + .write = probe_points_write, + .llseek = default_llseek, +}; + +static ssize_t probe_points_remove_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + size_t num_tkns; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + if (!num_tkns) { + ret = -EINVAL; + goto exit; + } + + ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_remove_fops = { + .open = simple_open, + .write = probe_points_remove_write, + .llseek = default_llseek, +}; + +static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev, + const char *name, mode_t mode, + const struct file_operations *fops) +{ + struct snd_sof_dfsentry *dfse; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->sdev = sdev; + + debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops); + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + return 0; +} +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) #define MAX_IPC_FLOOD_DURATION_MS 1000 #define MAX_IPC_FLOOD_COUNT 10000 @@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + err = snd_sof_debugfs_probe_item(sdev, "probe_points", + 0644, &probe_points_fops); + if (err < 0) + return err; + err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove", + 0200, &probe_points_remove_fops); + if (err < 0) + return err; +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) /* create read-write ipc_flood_count debugfs entry */ err = snd_sof_debugfs_buf_item(sdev, NULL, 0, diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index b2556f5e2871..b692752b2178 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -138,7 +138,7 @@ static int imx8_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) /* * DSP control. */ -static int imx8_run(struct snd_sof_dev *sdev) +static int imx8x_run(struct snd_sof_dev *sdev) { struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private; int ret; @@ -178,6 +178,24 @@ static int imx8_run(struct snd_sof_dev *sdev) return 0; } +static int imx8_run(struct snd_sof_dev *sdev) +{ + struct imx8_priv *dsp_priv = (struct imx8_priv *)sdev->private; + int ret; + + ret = imx_sc_misc_set_control(dsp_priv->sc_ipc, IMX_SC_R_DSP, + IMX_SC_C_OFS_SEL, 0); + if (ret < 0) { + dev_err(sdev->dev, "Error system address offset source select\n"); + return ret; + } + + imx_sc_pm_cpu_start(dsp_priv->sc_ipc, IMX_SC_R_DSP, true, + RESET_VECTOR_VADDR); + + return 0; +} + static int imx8_probe(struct snd_sof_dev *sdev) { struct platform_device *pdev = @@ -360,7 +378,7 @@ static struct snd_soc_dai_driver imx8_dai[] = { }, }; -/* i.MX8 ops */ +/* i.MX8 ops */ struct snd_sof_dsp_ops sof_imx8_ops = { /* probe and remove */ .probe = imx8_probe, @@ -390,6 +408,39 @@ struct snd_sof_dsp_ops sof_imx8_ops = { /* DAI drivers */ .drv = imx8_dai, .num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */ +}; +EXPORT_SYMBOL(sof_imx8_ops); + +/* i.MX8X ops */ +struct snd_sof_dsp_ops sof_imx8x_ops = { + /* probe and remove */ + .probe = imx8_probe, + .remove = imx8_remove, + /* DSP core boot */ + .run = imx8x_run, + + /* Block IO */ + .block_read = sof_block_read, + .block_write = sof_block_write, + + /* ipc */ + .send_msg = imx8_send_msg, + .fw_ready = sof_fw_ready, + .get_mailbox_offset = imx8_get_mailbox_offset, + .get_window_offset = imx8_get_window_offset, + + .ipc_msg_data = imx8_ipc_msg_data, + .ipc_pcm_params = imx8_ipc_pcm_params, + + /* module loading */ + .load_module = snd_sof_parse_module_memcpy, + .get_bar_index = imx8_get_bar_index, + /* firmware loading */ + .load_firmware = snd_sof_load_firmware_memcpy, + + /* DAI drivers */ + .drv = imx8_dai, + .num_drv = 1, /* we have only 1 ESAI interface on i.MX8 */ /* ALSA HW info flags */ .hw_info = SNDRV_PCM_INFO_MMAP | @@ -398,6 +449,6 @@ struct snd_sof_dsp_ops sof_imx8_ops = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; -EXPORT_SYMBOL(sof_imx8_ops); +EXPORT_SYMBOL(sof_imx8x_ops); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 56a837d2cb95..c9a2bee4b55c 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -305,6 +305,15 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC Say Y if you want to enable HDAudio codecs with SOF. If unsure select "N". +config SND_SOC_SOF_HDA_PROBES + bool "SOF enable probes over HDA" + depends on SND_SOC_SOF_DEBUG_PROBES + help + This option enables the data probing for Intel(R). + Intel(R) Skylake and newer platforms. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 bool "SOF enable DMI Link L1" help @@ -315,17 +324,6 @@ config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 Say Y if you want to enable DMI Link L1 If unsure, select "N". -config SND_SOC_SOF_HDA_COMMON_HDMI_CODEC - bool "SOF common HDA HDMI codec driver" - depends on SND_SOC_SOF_HDA_LINK - depends on SND_HDA_CODEC_HDMI - default SND_HDA_CODEC_HDMI - help - This adds support for HDMI audio by using the common HDA - HDMI/DisplayPort codec driver. - Say Y if you want to use the common codec driver with SOF. - If unsure select "Y". - endif ## SND_SOC_SOF_HDA_COMMON config SND_SOC_SOF_HDA_LINK_BASELINE diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index b8f58e006e29..cee02a2e00f4 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -9,6 +9,7 @@ snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-bus.o \ apl.o cnl.o +snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-compress.o snd-sof-intel-hda-objs := hda-codec.o diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 2483b15699e7..02218d22e51f 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -73,6 +73,15 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .pcm_trigger = hda_dsp_pcm_trigger, .pcm_pointer = hda_dsp_pcm_pointer, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) + /* probe callbacks */ + .probe_assign = hda_probe_compr_assign, + .probe_free = hda_probe_compr_free, + .probe_set_params = hda_probe_compr_set_params, + .probe_trigger = hda_probe_compr_trigger, + .probe_pointer = hda_probe_compr_pointer, +#endif + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 9e2d8afe0535..e427d00eca71 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -65,11 +65,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) hda_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, msg); - if (sdev->code_loading) { - sdev->code_loading = 0; - wake_up(&sdev->waitq); - } - cnl_ipc_dsp_done(sdev); spin_unlock_irq(&sdev->ipc_lock); @@ -171,23 +166,48 @@ static bool cnl_compact_ipc_compress(struct snd_sof_ipc_msg *msg, static int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) { + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + struct sof_ipc_cmd_hdr *hdr; u32 dr = 0; u32 dd = 0; + /* + * Currently the only compact IPC supported is the PM_GATE + * IPC which is used for transitioning the DSP between the + * D0I0 and D0I3 states. And these are sent only during the + * set_power_state() op. Therefore, there will never be a case + * that a compact IPC results in the DSP exiting D0I3 without + * the host and FW being in sync. + */ if (cnl_compact_ipc_compress(msg, &dr, &dd)) { /* send the message via IPC registers */ snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDD, dd); snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, CNL_DSP_REG_HIPCIDR_BUSY | dr); - } else { - /* send the message via mailbox */ - sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, - msg->msg_size); - snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, - CNL_DSP_REG_HIPCIDR_BUSY); + return 0; } + /* send the message via mailbox */ + sof_mailbox_write(sdev, sdev->host_box.offset, msg->msg_data, + msg->msg_size); + snd_sof_dsp_write(sdev, HDA_DSP_BAR, CNL_DSP_REG_HIPCIDR, + CNL_DSP_REG_HIPCIDR_BUSY); + + hdr = msg->msg_data; + + /* + * Use mod_delayed_work() to schedule the delayed work + * to avoid scheduling multiple workqueue items when + * IPCs are sent at a high-rate. mod_delayed_work() + * modifies the timer if the work is pending. + * Also, a new delayed work should not be queued after the + * the CTX_SAVE IPC, which is sent before the DSP enters D3. + */ + if (hdr->cmd != (SOF_IPC_GLB_PM_MSG | SOF_IPC_PM_CTX_SAVE)) + mod_delayed_work(system_wq, &hdev->d0i3_work, + msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS)); + return 0; } @@ -259,6 +279,15 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .pcm_trigger = hda_dsp_pcm_trigger, .pcm_pointer = hda_dsp_pcm_pointer, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) + /* probe callbacks */ + .probe_assign = hda_probe_compr_assign, + .probe_free = hda_probe_compr_free, + .probe_set_params = hda_probe_compr_set_params, + .probe_trigger = hda_probe_compr_trigger, + .probe_pointer = hda_probe_compr_pointer, +#endif + /* firmware loading */ .load_firmware = snd_sof_load_firmware_raw, diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index ff45075ef720..3041fbbb010a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -113,8 +113,14 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (ret < 0) return ret; - if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) + if ((resp & 0xFFFF0000) == IDISP_VID_INTEL) { + if (!hdev->bus->audio_component) { + dev_dbg(sdev->dev, + "iDisp hw present but no driver\n"); + return -ENOENT; + } hda_priv->need_display_power = true; + } /* * if common HDMI codec driver is not used, codec load @@ -203,6 +209,9 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); int ret; + if (!bus->audio_component) + return 0; + /* power down unconditionally */ snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); diff --git a/sound/soc/sof/intel/hda-compress.c b/sound/soc/sof/intel/hda-compress.c new file mode 100644 index 000000000000..38a1ebec8478 --- /dev/null +++ b/sound/soc/sof/intel/hda-compress.c @@ -0,0 +1,114 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski@intel.com> +// + +#include <sound/hdaudio_ext.h> +#include <sound/soc.h> +#include "../sof-priv.h" +#include "hda.h" + +static inline struct hdac_ext_stream * +hda_compr_get_stream(struct snd_compr_stream *cstream) +{ + return cstream->runtime->private_data; +} + +int hda_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream; + + stream = hda_dsp_stream_get(sdev, cstream->direction); + if (!stream) + return -EBUSY; + + hdac_stream(stream)->curr_pos = 0; + hdac_stream(stream)->cstream = cstream; + cstream->runtime->private_data = stream; + + return hdac_stream(stream)->stream_tag; +} + +int hda_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + int ret; + + ret = hda_dsp_stream_put(sdev, cstream->direction, + hdac_stream(stream)->stream_tag); + if (ret < 0) { + dev_dbg(sdev->dev, "stream put failed: %d\n", ret); + return ret; + } + + hdac_stream(stream)->cstream = NULL; + cstream->runtime->private_data = NULL; + + return 0; +} + +int hda_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct hdac_stream *hstream = hdac_stream(stream); + struct snd_dma_buffer *dmab; + u32 bits, rate; + int bps, ret; + + dmab = cstream->runtime->dma_buffer_p; + /* compr params do not store bit depth, default to S32_LE */ + bps = snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32_LE); + if (bps < 0) + return bps; + bits = hda_dsp_get_bits(sdev, bps); + rate = hda_dsp_get_mult_div(sdev, params->codec.sample_rate); + + hstream->format_val = rate | bits | (params->codec.ch_out - 1); + hstream->bufsize = cstream->runtime->buffer_size; + hstream->period_bytes = cstream->runtime->fragment_size; + hstream->no_period_wakeup = 0; + + ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL); + if (ret < 0) { + dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); + return ret; + } + + return 0; +} + +int hda_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + + return hda_dsp_stream_trigger(sdev, stream, cmd); +} + +int hda_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct snd_soc_pcm_stream *pstream; + + pstream = &dai->driver->capture; + tstamp->copied_total = hdac_stream(stream)->curr_pos; + tstamp->sampling_rate = snd_pcm_rate_bit_to_rate(pstream->rates); + + return 0; +} diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 871b71a15a63..6288b2f99540 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -18,6 +18,7 @@ #include <linux/module.h> #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <sound/hda_component.h> #include "../ops.h" #include "hda.h" @@ -64,15 +65,32 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); u32 cap, offset, feature; int count = 0; + int ret; + + /* + * On some devices, one reset cycle is necessary before reading + * capabilities + */ + ret = hda_dsp_ctrl_link_reset(sdev, true); + if (ret < 0) + return ret; + ret = hda_dsp_ctrl_link_reset(sdev, false); + if (ret < 0) + return ret; offset = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_LLCH); do { - cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset); - dev_dbg(sdev->dev, "checking for capabilities at offset 0x%x\n", offset & SOF_HDA_CAP_NEXT_MASK); + cap = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, offset); + + if (cap == -1) { + dev_dbg(bus->dev, "Invalid capability reg read\n"); + break; + } + feature = (cap & SOF_HDA_CAP_ID_MASK) >> SOF_HDA_CAP_ID_OFF; switch (feature) { @@ -105,8 +123,8 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) bus->mlcap = bus->remap_addr + offset; break; default: - dev_vdbg(sdev->dev, "found capability %d at 0x%x\n", - feature, offset); + dev_dbg(sdev->dev, "found capability %d at 0x%x\n", + feature, offset); break; } @@ -176,6 +194,9 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) if (bus->chip_init) return 0; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_set_codec_wakeup(bus, true); +#endif hda_dsp_ctrl_misc_clock_gating(sdev, false); if (full_reset) { @@ -183,7 +204,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) ret = hda_dsp_ctrl_link_reset(sdev, true); if (ret < 0) { dev_err(sdev->dev, "error: failed to reset HDA controller\n"); - return ret; + goto err; } usleep_range(500, 1000); @@ -192,7 +213,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) ret = hda_dsp_ctrl_link_reset(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); - return ret; + goto err; } usleep_range(1000, 1200); @@ -202,7 +223,8 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) /* check to see if controller is ready */ if (!snd_hdac_chip_readb(bus, GCTL)) { dev_dbg(bus->dev, "controller not ready!\n"); - return -EBUSY; + ret = -EBUSY; + goto err; } /* Accept unsolicited responses */ @@ -268,7 +290,11 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset) bus->chip_init = true; +err: hda_dsp_ctrl_misc_clock_gating(sdev, true); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + snd_hdac_set_codec_wakeup(bus, false); +#endif return ret; } diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9c6e3f990ee3..833dc303b394 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -204,7 +204,7 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = hstream->bus; struct hdac_ext_stream *link_dev; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct sof_intel_hda_stream *hda_stream; struct hda_pipe_params p_params = {0}; struct hdac_ext_link *link; @@ -293,7 +293,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, bus = hstream->bus; rtd = snd_pcm_substream_chip(substream); - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; @@ -374,7 +374,7 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, rtd->codec_dai->component->name); + link = snd_hdac_ext_bus_get_link(bus, asoc_rtd_to_codec(rtd, 0)->component->name); if (!link) return -EINVAL; @@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = { .trigger = hda_link_pcm_trigger, .prepare = hda_link_pcm_prepare, }; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#include "../compress.h" + +static struct snd_soc_cdai_ops sof_probe_compr_ops = { + .startup = sof_probe_compr_open, + .shutdown = sof_probe_compr_free, + .set_params = sof_probe_compr_set_params, + .trigger = sof_probe_compr_trigger, + .pointer = sof_probe_compr_pointer, +}; + +#endif #endif /* @@ -409,56 +422,167 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = { struct snd_soc_dai_driver skl_dai[] = { { .name = "SSP0 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP1 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP2 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP3 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP4 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "SSP5 Pin", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "DMIC01 Pin", + .capture = { + .channels_min = 1, + .channels_max = 4, + }, }, { .name = "DMIC16k Pin", + .capture = { + .channels_min = 1, + .channels_max = 4, + }, }, #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) { .name = "iDisp1 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp2 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp3 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "iDisp4 Pin", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "Analog CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, }, { .name = "Digital CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, }, { .name = "Alt Analog CPU DAI", .ops = &hda_link_dai_ops, + .playback = { + .channels_min = 1, + .channels_max = 16, + }, + .capture = { + .channels_min = 1, + .channels_max = 16, + }, +}, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +{ + .name = "Probe Extraction CPU DAI", + .compress_new = snd_soc_new_compress, + .cops = &sof_probe_compr_ops, + .capture = { + .stream_name = "Probe Extraction", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + }, }, #endif +#endif }; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 0848b79967a9..99087b6afb67 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -15,12 +15,21 @@ * Hardware interface for generic Intel audio DSP HDA IP */ +#include <linux/module.h> #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" #include "hda-ipc.h" +static bool hda_enable_trace_D0I3_S0; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG) +module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444); +MODULE_PARM_DESC(enable_trace_D0I3_S0, + "SOF HDA enable trace when the DSP is in D0I3 in S0"); +#endif + /* * DSP Core control. */ @@ -334,17 +343,15 @@ static int hda_dsp_send_pm_gate_ipc(struct snd_sof_dev *sdev, u32 flags) pm_gate.flags = flags; /* send pm_gate ipc to dsp */ - return sof_ipc_tx_message(sdev->ipc, pm_gate.hdr.cmd, &pm_gate, - sizeof(pm_gate), &reply, sizeof(reply)); + return sof_ipc_tx_message_no_pm(sdev->ipc, pm_gate.hdr.cmd, + &pm_gate, sizeof(pm_gate), &reply, + sizeof(reply)); } -int hda_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate) +static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) { struct hdac_bus *bus = sof_to_bus(sdev); - u32 flags; int ret; - u8 value; /* Write to D0I3C after Command-In-Progress bit is cleared */ ret = hda_dsp_wait_d0i3c_done(sdev); @@ -354,7 +361,6 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, } /* Update D0I3C register */ - value = d0_substate == SOF_DSP_D0I3 ? SOF_HDA_VS_D0I3C_I3 : 0; snd_hdac_chip_updateb(bus, VS_D0I3C, SOF_HDA_VS_D0I3C_I3, value); /* Wait for cmd in progress to be cleared before exiting the function */ @@ -367,20 +373,218 @@ int hda_dsp_set_power_state(struct snd_sof_dev *sdev, dev_vdbg(bus->dev, "D0I3C updated, register = 0x%x\n", snd_hdac_chip_readb(bus, VS_D0I3C)); - if (d0_substate == SOF_DSP_D0I0) - flags = HDA_PM_PPG;/* prevent power gating in D0 */ - else - flags = HDA_PM_NO_DMA_TRACE;/* disable DMA trace in D0I3*/ + return 0; +} - /* sending pm_gate IPC */ - ret = hda_dsp_send_pm_gate_ipc(sdev, flags); +static int hda_dsp_set_D0_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + u32 flags = 0; + int ret; + u8 value = 0; + + /* + * Sanity check for illegal state transitions + * The only allowed transitions are: + * 1. D3 -> D0I0 + * 2. D0I0 -> D0I3 + * 3. D0I3 -> D0I0 + */ + switch (sdev->dsp_power_state.state) { + case SOF_DSP_PM_D0: + /* Follow the sequence below for D0 substate transitions */ + break; + case SOF_DSP_PM_D3: + /* Follow regular flow for D3 -> D0 transition */ + return 0; + default: + dev_err(sdev->dev, "error: transition from %d to %d not allowed\n", + sdev->dsp_power_state.state, target_state->state); + return -EINVAL; + } + + /* Set flags and register value for D0 target substate */ + if (target_state->substate == SOF_HDA_DSP_PM_D0I3) { + value = SOF_HDA_VS_D0I3C_I3; + + /* + * Trace DMA is disabled by default when the DSP enters D0I3. + * But it can be kept enabled when the DSP enters D0I3 while the + * system is in S0 for debug. + */ + if (hda_enable_trace_D0I3_S0 && + sdev->system_suspend_target != SOF_SUSPEND_NONE) + flags = HDA_PM_NO_DMA_TRACE; + } else { + /* prevent power gating in D0I0 */ + flags = HDA_PM_PPG; + } + + /* update D0I3C register */ + ret = hda_dsp_update_d0i3c_register(sdev, value); if (ret < 0) + return ret; + + /* + * Notify the DSP of the state change. + * If this IPC fails, revert the D0I3C register update in order + * to prevent partial state change. + */ + ret = hda_dsp_send_pm_gate_ipc(sdev, flags); + if (ret < 0) { dev_err(sdev->dev, "error: PM_GATE ipc error %d\n", ret); + goto revert; + } + + return ret; + +revert: + /* fallback to the previous register value */ + value = value ? 0 : SOF_HDA_VS_D0I3C_I3; + + /* + * This can fail but return the IPC error to signal that + * the state change failed. + */ + hda_dsp_update_d0i3c_register(sdev, value); return ret; } +/* helper to log DSP state */ +static void hda_dsp_state_log(struct snd_sof_dev *sdev) +{ + switch (sdev->dsp_power_state.state) { + case SOF_DSP_PM_D0: + switch (sdev->dsp_power_state.substate) { + case SOF_HDA_DSP_PM_D0I0: + dev_dbg(sdev->dev, "Current DSP power state: D0I0\n"); + break; + case SOF_HDA_DSP_PM_D0I3: + dev_dbg(sdev->dev, "Current DSP power state: D0I3\n"); + break; + default: + dev_dbg(sdev->dev, "Unknown DSP D0 substate: %d\n", + sdev->dsp_power_state.substate); + break; + } + break; + case SOF_DSP_PM_D1: + dev_dbg(sdev->dev, "Current DSP power state: D1\n"); + break; + case SOF_DSP_PM_D2: + dev_dbg(sdev->dev, "Current DSP power state: D2\n"); + break; + case SOF_DSP_PM_D3_HOT: + dev_dbg(sdev->dev, "Current DSP power state: D3_HOT\n"); + break; + case SOF_DSP_PM_D3: + dev_dbg(sdev->dev, "Current DSP power state: D3\n"); + break; + case SOF_DSP_PM_D3_COLD: + dev_dbg(sdev->dev, "Current DSP power state: D3_COLD\n"); + break; + default: + dev_dbg(sdev->dev, "Unknown DSP power state: %d\n", + sdev->dsp_power_state.state); + break; + } +} + +/* + * All DSP power state transitions are initiated by the driver. + * If the requested state change fails, the error is simply returned. + * Further state transitions are attempted only when the set_power_save() op + * is called again either because of a new IPC sent to the DSP or + * during system suspend/resume. + */ +int hda_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) +{ + int ret = 0; + + /* + * When the DSP is already in D0I3 and the target state is D0I3, + * it could be the case that the DSP is in D0I3 during S0 + * and the system is suspending to S0Ix. Therefore, + * hda_dsp_set_D0_state() must be called to disable trace DMA + * by sending the PM_GATE IPC to the FW. + */ + if (target_state->substate == SOF_HDA_DSP_PM_D0I3 && + sdev->system_suspend_target == SOF_SUSPEND_S0IX) + goto set_state; + + /* + * For all other cases, return without doing anything if + * the DSP is already in the target state. + */ + if (target_state->state == sdev->dsp_power_state.state && + target_state->substate == sdev->dsp_power_state.substate) + return 0; + +set_state: + switch (target_state->state) { + case SOF_DSP_PM_D0: + ret = hda_dsp_set_D0_state(sdev, target_state); + break; + case SOF_DSP_PM_D3: + /* The only allowed transition is: D0I0 -> D3 */ + if (sdev->dsp_power_state.state == SOF_DSP_PM_D0 && + sdev->dsp_power_state.substate == SOF_HDA_DSP_PM_D0I0) + break; + + dev_err(sdev->dev, + "error: transition from %d to %d not allowed\n", + sdev->dsp_power_state.state, target_state->state); + return -EINVAL; + default: + dev_err(sdev->dev, "error: target state unsupported %d\n", + target_state->state); + return -EINVAL; + } + if (ret < 0) { + dev_err(sdev->dev, + "failed to set requested target DSP state %d substate %d\n", + target_state->state, target_state->substate); + return ret; + } + + sdev->dsp_power_state = *target_state; + hda_dsp_state_log(sdev); + return ret; +} + +/* + * Audio DSP states may transform as below:- + * + * Opportunistic D0I3 in S0 + * Runtime +---------------------+ Delayed D0i3 work timeout + * suspend | +--------------------+ + * +------------+ D0I0(active) | | + * | | <---------------+ | + * | +--------> | New IPC | | + * | |Runtime +--^--+---------^--+--+ (via mailbox) | | + * | |resume | | | | | | + * | | | | | | | | + * | | System| | | | | | + * | | resume| | S3/S0IX | | | | + * | | | | suspend | | S0IX | | + * | | | | | |suspend | | + * | | | | | | | | + * | | | | | | | | + * +-v---+-----------+--v-------+ | | +------+----v----+ + * | | | +-----------> | + * | D3 (suspended) | | | D0I3 | + * | | +--------------+ | + * | | System resume | | + * +----------------------------+ +----------------+ + * + * S0IX suspend: The DSP is in D0I3 if any D0I3-compatible streams + * ignored the suspend trigger. Otherwise the DSP + * is in D3. + */ + static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; @@ -390,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); @@ -486,10 +692,24 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + .substate = SOF_HDA_DSP_PM_D0I0, + }; + int ret; - if (sdev->s0_suspend) { + /* resume from D0I3 */ + if (sdev->dsp_power_state.state == SOF_DSP_PM_D0) { hda_codec_i915_display_power(sdev, true); + /* Set DSP power state */ + ret = snd_sof_dsp_set_power_state(sdev, &target_state); + if (ret < 0) { + dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", + target_state.state, target_state.substate); + return ret; + } + /* restore L1SEN bit */ if (hda->l1_support_changed) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -503,13 +723,26 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) } /* init hda controller. DSP cores will be powered up during fw boot */ - return hda_resume(sdev, false); + ret = hda_resume(sdev, false); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; + int ret; + /* init hda controller. DSP cores will be powered up during fw boot */ - return hda_resume(sdev, true); + ret = hda_resume(sdev, true); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -527,21 +760,47 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D3, + }; + int ret; + /* stop hda controller and power dsp off */ - return hda_suspend(sdev, true); + ret = hda_suspend(sdev, true); + if (ret < 0) + return ret; + + return snd_sof_dsp_set_power_state(sdev, &target_state); } -int hda_dsp_suspend(struct snd_sof_dev *sdev) +int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct hdac_bus *bus = sof_to_bus(sdev); struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_dsp_power_state target_dsp_state = { + .state = target_state, + .substate = target_state == SOF_DSP_PM_D0 ? + SOF_HDA_DSP_PM_D0I3 : 0, + }; int ret; - if (sdev->s0_suspend) { + /* cancel any attempt for DSP D0I3 */ + cancel_delayed_work_sync(&hda->d0i3_work); + + if (target_state == SOF_DSP_PM_D0) { /* we can't keep a wakeref to display driver at suspend */ hda_codec_i915_display_power(sdev, false); + /* Set DSP power state */ + ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); + if (ret < 0) { + dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", + target_dsp_state.state, + target_dsp_state.substate); + return ret; + } + /* enable L1SEN to make sure the system can enter S0Ix */ hda->l1_support_changed = snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -562,7 +821,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev) return ret; } - return 0; + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) @@ -588,7 +847,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) */ if (stream->link_substream) { rtd = snd_pcm_substream_chip(stream->link_substream); - name = rtd->codec_dai->component->name; + name = asoc_rtd_to_codec(rtd, 0)->component->name; link = snd_hdac_ext_bus_get_link(bus, name); if (!link) return -EINVAL; @@ -606,3 +865,33 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) #endif return 0; } + +void hda_dsp_d0i3_work(struct work_struct *work) +{ + struct sof_intel_hda_dev *hdev = container_of(work, + struct sof_intel_hda_dev, + d0i3_work.work); + struct hdac_bus *bus = &hdev->hbus.core; + struct snd_sof_dev *sdev = dev_get_drvdata(bus->dev); + struct sof_dsp_power_state target_state; + int ret; + + target_state.state = SOF_DSP_PM_D0; + + /* DSP can enter D0I3 iff only D0I3-compatible streams are active */ + if (snd_sof_dsp_only_d0i3_compatible_stream_active(sdev)) + target_state.substate = SOF_HDA_DSP_PM_D0I3; + else + target_state.substate = SOF_HDA_DSP_PM_D0I0; + + /* remain in D0I0 */ + if (target_state.substate == SOF_HDA_DSP_PM_D0I0) + return; + + /* This can fail but error cannot be propagated */ + ret = snd_sof_dsp_set_power_state(sdev, &target_state); + if (ret < 0) + dev_err_ratelimited(sdev->dev, + "error: failed to set DSP state %d substate %d\n", + target_state.state, target_state.substate); +} diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 1837f66e361f..6062bb6011fb 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) ret = reply.error; } else { /* reply correct size ? */ - if (reply.hdr.size != msg->reply_size) { + if (reply.hdr.size != msg->reply_size && + /* getter payload is never known upfront */ + !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; @@ -123,12 +125,6 @@ out: } -static bool hda_dsp_ipc_is_sof(uint32_t msg) -{ - return (msg & (HDA_DSP_IPC_PURGE_FW | 0xf << 9)) != msg || - (msg & HDA_DSP_IPC_PURGE_FW) != HDA_DSP_IPC_PURGE_FW; -} - /* IPC handler thread */ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) { @@ -174,17 +170,9 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) */ spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core - ignore ROM messages */ - if (hda_dsp_ipc_is_sof(msg)) { - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); - } - - /* wake up sleeper if we are loading code */ - if (sdev->code_loading) { - sdev->code_loading = 0; - wake_up(&sdev->waitq); - } + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); /* set the done bit */ hda_dsp_ipc_dsp_done(sdev); diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 8852184a2569..e1550ccd0a49 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -131,6 +131,12 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, const void *fwdata, goto err; } + /* set DONE bit to clear the reply IPC message */ + snd_sof_dsp_update_bits_forced(sdev, HDA_DSP_BAR, + chip->ipc_ack, + chip->ipc_ack_mask, + chip->ipc_ack_mask); + /* step 5: power down corex */ ret = hda_dsp_core_power_down(sdev, chip->cores_mask & ~(HDA_DSP_CORE_MASK(0))); @@ -173,9 +179,6 @@ static int cl_trigger(struct snd_sof_dev *sdev, /* code loader is special case that reuses stream ops */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: - wait_event_timeout(sdev->waitq, !sdev->code_loading, - HDA_DSP_CL_TRIGGER_TIMEOUT); - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, 1 << hstream->index, 1 << hstream->index); @@ -344,6 +347,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) } /* + * When a SoundWire link is in clock stop state, a Slave + * device may trigger in-band wakes for events such as jack + * insertion or acoustic event detection. This event will lead + * to a WAKEEN interrupt, handled by the PCI device and routed + * to PME if the PCI device is in D3. The resume function in + * audio PCI driver will be invoked by ACPI for PME event and + * initialize the device and process WAKEEN interrupt. + * + * The WAKEEN interrupt should be processed ASAP to prevent an + * interrupt flood, otherwise other interrupts, such IPC, + * cannot work normally. The WAKEEN is handled after the ROM + * is initialized successfully, which ensures power rails are + * enabled before accessing the SoundWire SHIM registers + */ + if (!sdev->first_boot) + hda_sdw_process_wakeen(sdev); + + /* * at this point DSP ROM has been initialized and * should be ready for code loading and firmware boot */ @@ -396,6 +417,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 23872f6e708d..a46a6baa1c3f 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -27,7 +27,7 @@ #define SDnFMT_BITS(x) ((x) << 4) #define SDnFMT_CHAN(x) ((x) << 0) -static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate) +u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate) { switch (rate) { case 8000: @@ -61,7 +61,7 @@ static inline u32 get_mult_div(struct snd_sof_dev *sdev, int rate) } }; -static inline u32 get_bits(struct snd_sof_dev *sdev, int sample_bits) +u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits) { switch (sample_bits) { case 8: @@ -95,8 +95,8 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, u32 size, rate, bits; size = params_buffer_bytes(params); - rate = get_mult_div(sdev, params_rate(params)); - bits = get_bits(sdev, params_width(params)); + rate = hda_dsp_get_mult_div(sdev, params_rate(params)); + bits = hda_dsp_get_bits(sdev, params_width(params)); hstream->substream = substream; diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index c0ab9bb2a797..5d386956906f 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -547,6 +547,8 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); + stream->substream = NULL; + return 0; } @@ -571,6 +573,22 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) return ret; } +static void +hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +{ + u64 prev_pos, pos, num_bytes; + + div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + + if (pos < prev_pos) + num_bytes = (buffer_size - prev_pos) + pos; + else + num_bytes = pos - prev_pos; + + hstream->curr_pos += num_bytes; +} + static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) { struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); @@ -588,14 +606,19 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) snd_hdac_stream_writeb(s, SD_STS, sd_status); active = true; - if (!s->substream || + if ((!s->substream && !s->cstream) || !s->running || (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) continue; /* Inform ALSA only in case not do that with IPC */ - if (sof_hda->no_ipc_position) + if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); + } else if (s->cstream) { + hda_dsp_set_bytes_transferred(s, + s->cstream->runtime->buffer_size); + snd_compr_fragment_elapsed(s->cstream); + } } } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 25946a1c2822..211e91e79eae 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,10 +18,14 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <linux/acpi.h> #include <linux/module.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -34,6 +38,235 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +/* + * The default for SoundWire clock stop quirks is to power gate the IP + * and do a Bus Reset, this will need to be modified when the DSP + * needs to remain in D0i3 so that the Master does not lose context + * and enumeration is not required on clock restart + */ +static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET; +module_param(sdw_clock_stop_quirks, int, 0444); +MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks"); + +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + void *sdw; + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; + res.clock_stop_quirks = sdw_clock_stop_quirks; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return 0; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + +static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (hdev->sdw && + snd_sof_dsp_read(sdev, HDA_DSP_BAR, + HDA_DSP_REG_SNDW_WAKE_STS)) + return true; + + return false; +} + +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return; + + sdw_intel_process_wakeen_event(hdev->sdw); +} + +#endif + /* * Debug */ @@ -54,8 +287,7 @@ static int hda_dmic_num = -1; module_param_named(dmic_num, hda_dmic_num, int, 0444); MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number"); -static bool hda_codec_use_common_hdmi = - IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_COMMON_HDMI_CODEC); +static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI); module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444); MODULE_PARM_DESC(use_common_hdmi, "SOF HDA use common HDMI codec driver"); #endif @@ -288,10 +520,8 @@ static int hda_init(struct snd_sof_dev *sdev) /* init i915 and HDMI codecs */ ret = hda_codec_i915_init(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: init i915 and HDMI codec failed\n"); - return ret; - } + if (ret < 0) + dev_warn(sdev->dev, "init of i915 and HDMI codec failed\n"); /* get controller capabilities */ ret = hda_dsp_ctrl_get_caps(sdev); @@ -349,9 +579,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,12 +598,37 @@ static int hda_init_caps(struct snd_sof_dev *sdev) if (ret < 0) { dev_err(bus->dev, "error: init chip failed with ret: %d\n", ret); -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - hda_codec_i915_exit(sdev); -#endif return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_dbg(sdev->dev, "skipping SoundWire, ACPI scan error\n"); + goto skip_soundwire; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + dev_dbg(sdev->dev, "skipping SoundWire, no links enabled\n"); + goto skip_soundwire; + } + + /* + * probe/allocate SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + * It's entirely possible to have a mix of I2S/DMIC/SoundWire + * devices, so we allocate the resources in all cases. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + +skip_soundwire: + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -379,7 +637,7 @@ static int hda_init_caps(struct snd_sof_dev *sdev) hda_codec_probe_bus(sdev, hda_codec_use_common_hdmi); if (!HDA_IDISP_CODEC(bus->codec_mask)) - hda_codec_i915_exit(sdev); + hda_codec_i915_display_power(sdev, false); /* * we are done probing so decrement link counts @@ -427,6 +685,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -435,6 +694,12 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + + if (hda_sdw_check_wakeen_irq(sdev)) + hda_sdw_process_wakeen(sdev); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, @@ -590,12 +855,11 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) hda_dsp_ctrl_ppcap_enable(sdev, true); hda_dsp_ctrl_ppcap_int_enable(sdev, true); - /* initialize waitq for code loading */ - init_waitqueue_head(&sdev->waitq); - /* set default mailbox offset for FW ready message */ sdev->dsp_box.offset = HDA_DSP_MBOX_UPLINK_OFFSET; + INIT_DELAYED_WORK(&hdev->d0i3_work, hda_dsp_d0i3_work); + return 0; free_ipc_irq: @@ -621,11 +885,16 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) struct pci_dev *pci = to_pci_dev(sdev->dev); const struct sof_intel_dsp_desc *chip = hda->desc; + /* cancel any attempt for DSP D0I3 */ + cancel_delayed_work_sync(&hda->d0i3_work); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* codec removal, invoke bus_device_remove */ snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); @@ -694,12 +963,11 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) /* * If no machine driver is found, then: * - * hda machine driver is used if : - * 1. there is one HDMI codec and one external HDAudio codec - * 2. only HDMI codec + * generic hda machine driver can handle: + * - one HDMI codec, and/or + * - one external HDAudio codec */ - if (!pdata->machine && codec_num <= 2 && - HDA_IDISP_CODEC(bus->codec_mask)) { + if (!pdata->machine && codec_num <= 2) { hda_mach = snd_soc_acpi_intel_hda_machines; /* topology: use the info from hda_machines */ @@ -709,7 +977,7 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) dev_info(bus->dev, "using HDA machine driver %s now\n", hda_mach->drv_name); - if (codec_num == 1) + if (codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask)) idisp_str = "-idisp"; else idisp_str = ""; @@ -763,6 +1031,123 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) } #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* Check if all Slaves defined on the link can be found */ +static bool link_slaves_found(struct snd_sof_dev *sdev, + const struct snd_soc_acpi_link_adr *link, + struct sdw_intel_ctx *sdw) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + struct sdw_intel_slave_id *ids = sdw->ids; + int num_slaves = sdw->num_slaves; + unsigned int part_id, link_id, unique_id, mfg_id; + int i, j; + + for (i = 0; i < link->num_adr; i++) { + u64 adr = link->adr_d[i].adr; + + mfg_id = SDW_MFG_ID(adr); + part_id = SDW_PART_ID(adr); + link_id = SDW_DISCO_LINK_ID(adr); + for (j = 0; j < num_slaves; j++) { + if (ids[j].link_id != link_id || + ids[j].id.part_id != part_id || + ids[j].id.mfg_id != mfg_id) + continue; + /* + * we have to check unique id + * if there is more than one + * Slave on the link + */ + unique_id = SDW_UNIQUE_ID(adr); + if (link->num_adr == 1 || + ids[j].id.unique_id == SDW_IGNORED_UNIQUE_ID || + ids[j].id.unique_id == unique_id) { + dev_dbg(bus->dev, + "found %x at link %d\n", + part_id, link_id); + break; + } + } + if (j == num_slaves) { + dev_dbg(bus->dev, + "Slave %x not found\n", + part_id); + return false; + } + } + return true; +} + +static int hda_sdw_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *pdata = sdev->pdata; + const struct snd_soc_acpi_link_adr *link; + struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_soc_acpi_mach *mach; + struct sof_intel_hda_dev *hdev; + u32 link_mask; + int i; + + hdev = pdata->hw_pdata; + link_mask = hdev->info.link_mask; + + /* + * Select SoundWire machine driver if needed using the + * alternate tables. This case deals with SoundWire-only + * machines, for mixed cases with I2C/I2S the detection relies + * on the HID list. + */ + if (link_mask && !pdata->machine) { + for (mach = pdata->desc->alt_machines; + mach && mach->link_mask; mach++) { + if (mach->link_mask != link_mask) + continue; + + /* No need to match adr if there is no links defined */ + if (!mach->links) + break; + + link = mach->links; + for (i = 0; i < hdev->info.count && link->num_adr; + i++, link++) { + /* + * Try next machine if any expected Slaves + * are not found on this link. + */ + if (!link_slaves_found(sdev, link, hdev->sdw)) + break; + } + /* Found if all Slaves are checked */ + if (i == hdev->info.count || !link->num_adr) + break; + } + if (mach && mach->link_mask) { + dev_dbg(bus->dev, + "SoundWire machine driver %s topology %s\n", + mach->drv_name, + mach->sof_tplg_filename); + pdata->machine = mach; + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.platform = dev_name(sdev->dev); + pdata->fw_filename = mach->sof_fw_filename; + pdata->tplg_filename = mach->sof_tplg_filename; + } else { + dev_info(sdev->dev, + "No SoundWire machine driver found\n"); + } + } + + return 0; +} +#else +static int hda_sdw_machine_select(struct snd_sof_dev *sdev) +{ + return 0; +} +#endif + void hda_set_mach_params(const struct snd_soc_acpi_mach *mach, struct device *dev) { @@ -782,9 +1167,19 @@ void hda_machine_select(struct snd_sof_dev *sdev) if (mach) { sof_pdata->tplg_filename = mach->sof_tplg_filename; sof_pdata->machine = mach; + + if (mach->link_mask) { + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + } } /* + * If I2S fails, try SoundWire + */ + hda_sdw_machine_select(sdev); + + /* * Choose HDA generic machine driver if mach is NULL. * Otherwise, set certain mach params. */ diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 6191d9192fae..e9825798de77 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,9 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> +#include <sound/compress_driver.h> #include <sound/hda_codec.h> #include <sound/hdaudio_ext.h> #include "shim.h" @@ -174,7 +177,6 @@ * value cannot be read back within the specified time. */ #define HDA_DSP_STREAM_RUN_TIMEOUT 300 -#define HDA_DSP_CL_TRIGGER_TIMEOUT 300 #define HDA_DSP_SPIB_ENABLE 1 #define HDA_DSP_SPIB_DISABLE 0 @@ -230,6 +232,9 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) +#define HDA_DSP_REG_SNDW_WAKE_STS 0x2C192 + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -348,7 +353,13 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#define SOF_SKL_NUM_DAIS 16 +#else #define SOF_SKL_NUM_DAIS 15 +#endif + #else #define SOF_SKL_NUM_DAIS 8 #endif @@ -392,6 +403,19 @@ struct sof_intel_dsp_bdl { #define SOF_HDA_PLAYBACK 0 #define SOF_HDA_CAPTURE 1 +/* + * Time in ms for opportunistic D0I3 entry delay. + * This has been deliberately chosen to be long to avoid race conditions. + * Could be optimized in future. + */ +#define SOF_HDA_D0I3_WORK_DELAY_MS 5000 + +/* HDA DSP D0 substate */ +enum sof_hda_D0_substate { + SOF_HDA_DSP_PM_D0I0, /* default D0 substate */ + SOF_HDA_DSP_PM_D0I3, /* low power D0 substate */ +}; + /* represents DSP HDA controller frontend - i.e. host facing control */ struct sof_intel_hda_dev { @@ -414,6 +438,15 @@ struct sof_intel_hda_dev { /* DMIC device */ struct platform_device *dmic_dev; + + /* delayed work to enter D0I3 opportunistically */ + struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -469,9 +502,9 @@ void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev); void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev); int hda_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); + const struct sof_dsp_power_state *target_state); -int hda_dsp_suspend(struct snd_sof_dev *sdev); +int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state); int hda_dsp_resume(struct snd_sof_dev *sdev); int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev); int hda_dsp_runtime_resume(struct snd_sof_dev *sdev); @@ -481,10 +514,13 @@ void hda_dsp_dump_skl(struct snd_sof_dev *sdev, u32 flags); void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags); void hda_ipc_dump(struct snd_sof_dev *sdev); void hda_ipc_irq_dump(struct snd_sof_dev *sdev); +void hda_dsp_d0i3_work(struct work_struct *work); /* * DSP PCM Operations. */ +u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate); +u32 hda_dsp_get_bits(struct snd_sof_dev *sdev, int sample_bits); int hda_dsp_pcm_open(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); int hda_dsp_pcm_close(struct snd_sof_dev *sdev, @@ -533,6 +569,29 @@ int hda_ipc_pcm_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, const struct sof_ipc_pcm_params_reply *reply); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +/* + * Probe Compress Operations. + */ +int hda_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int hda_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int hda_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); +int hda_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int hda_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); +#endif + /* * DSP IPC Operations. */ @@ -606,6 +665,61 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} + +static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ +} +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 78aa1da7c7a9..1c6794918cbb 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -214,15 +214,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, snd_sof_handle_fw_exception(ipc->sdev); ret = -ETIMEDOUT; } else { - /* copy the data returned from DSP */ ret = msg->reply_error; - if (msg->reply_size) - memcpy(reply_data, msg->reply_data, msg->reply_size); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", hdr->cmd, msg->reply_size); - else + } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (msg->reply_size) + /* copy the data returned from DSP */ + memcpy(reply_data, msg->reply_data, + msg->reply_size); + } } return ret; @@ -268,7 +270,6 @@ static int sof_ipc_tx_message_unlocked(struct snd_sof_ipc *ipc, u32 header, spin_unlock_irq(&sdev->ipc_lock); if (ret < 0) { - /* So far IPC TX never fails, consider making the above void */ dev_err_ratelimited(sdev->dev, "error: ipc tx failed with error %d\n", ret); @@ -289,6 +290,32 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, void *msg_data, size_t msg_bytes, void *reply_data, size_t reply_bytes) { + const struct sof_dsp_power_state target_state = { + .state = SOF_DSP_PM_D0, + }; + int ret; + + /* ensure the DSP is in D0 before sending a new IPC */ + ret = snd_sof_dsp_set_power_state(ipc->sdev, &target_state); + if (ret < 0) { + dev_err(ipc->sdev->dev, "error: resuming DSP %d\n", ret); + return ret; + } + + return sof_ipc_tx_message_no_pm(ipc, header, msg_data, msg_bytes, + reply_data, reply_bytes); +} +EXPORT_SYMBOL(sof_ipc_tx_message); + +/* + * send IPC message from host to DSP without modifying the DSP state. + * This will be used for IPC's that can be handled by the DSP + * even in a low-power D0 substate. + */ +int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, + void *reply_data, size_t reply_bytes) +{ int ret; if (msg_bytes > SOF_IPC_MSG_MAX_SIZE || @@ -305,7 +332,7 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, return ret; } -EXPORT_SYMBOL(sof_ipc_tx_message); +EXPORT_SYMBOL(sof_ipc_tx_message_no_pm); /* handle reply message from DSP */ int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index fc4ab51bacf4..1f2e0be812bd 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -95,9 +95,6 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) /* process structure data */ switch (ext_hdr->type) { - case SOF_IPC_EXT_DMA_BUFFER: - ret = 0; - break; case SOF_IPC_EXT_WINDOW: ret = get_ext_windows(sdev, ext_hdr); break; @@ -469,9 +466,6 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) const char *fw_filename; int ret; - /* set code loading condition to true */ - sdev->code_loading = 1; - /* Don't request firmware again if firmware is already requested */ if (plat_data->fw) return 0; diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index e929a6e0058f..a771500ac442 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -146,10 +146,11 @@ static inline int snd_sof_dsp_resume(struct snd_sof_dev *sdev) return 0; } -static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev) +static inline int snd_sof_dsp_suspend(struct snd_sof_dev *sdev, + u32 target_state) { if (sof_ops(sdev)->suspend) - return sof_ops(sdev)->suspend(sdev); + return sof_ops(sdev)->suspend(sdev, target_state); return 0; } @@ -193,14 +194,15 @@ static inline int snd_sof_dsp_set_clk(struct snd_sof_dev *sdev, u32 freq) return 0; } -static inline int snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, - enum sof_d0_substate substate) +static inline int +snd_sof_dsp_set_power_state(struct snd_sof_dev *sdev, + const struct sof_dsp_power_state *target_state) { if (sof_ops(sdev)->set_power_state) - return sof_ops(sdev)->set_power_state(sdev, substate); + return sof_ops(sdev)->set_power_state(sdev, target_state); - /* D0 substate is not supported */ - return -ENOTSUPP; + /* D0 substate is not supported, do nothing here. */ + return 0; } /* debug */ @@ -391,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev, return 0; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +static inline int +snd_sof_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_assign(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_free(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai); +} + +static inline int +snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai); +} + +static inline int +snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer) + return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai); + + return 0; +} +#endif + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 29435ba2d329..47cd741f2a8c 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -16,6 +16,9 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "compress.h" +#endif /* Create DMA buffer page table for DSP */ static int create_page_table(struct snd_soc_component *component, @@ -54,7 +57,7 @@ static int sof_pcm_dsp_params(struct snd_sof_pcm *spcm, struct snd_pcm_substream /* * sof pcm period elapse work */ -static void sof_pcm_period_elapsed_work(struct work_struct *work) +void snd_sof_pcm_period_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = container_of(work, struct snd_sof_pcm_stream, @@ -372,7 +375,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component, stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; break; case SNDRV_PCM_TRIGGER_SUSPEND: - if (sdev->s0_suspend && + if (sdev->system_suspend_target == SOF_SUSPEND_S0IX && spcm->stream[substream->stream].d0i3_compatible) { /* * trap the event, not sending trigger stop to @@ -472,8 +475,6 @@ static int sof_pcm_open(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: open stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - INIT_WORK(&spcm->stream[substream->stream].period_elapsed_work, - sof_pcm_period_elapsed_work); caps = &spcm->pcm.caps[substream->stream]; @@ -598,8 +599,7 @@ static int sof_pcm_new(struct snd_soc_component *component, snd_pcm_set_managed_buffer(pcm->streams[stream].substream, SNDRV_DMA_TYPE_DEV_SG, sdev->dev, - le32_to_cpu(caps->buffer_size_min), - le32_to_cpu(caps->buffer_size_max)); + 0, le32_to_cpu(caps->buffer_size_max)); capture: stream = SNDRV_PCM_STREAM_CAPTURE; @@ -621,8 +621,7 @@ capture: snd_pcm_set_managed_buffer(pcm->streams[stream].substream, SNDRV_DMA_TYPE_DEV_SG, sdev->dev, - le32_to_cpu(caps->buffer_size_min), - le32_to_cpu(caps->buffer_size_max)); + 0, le32_to_cpu(caps->buffer_size_max)); return 0; } @@ -788,6 +787,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) pd->compr_ops = &sof_compressed_ops; #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* override cops when probe support is enabled */ + pd->compr_ops = &sof_probe_compressed_ops; +#endif pd->pcm_construct = sof_pcm_new; pd->ignore_machine = drv_name; pd->be_hw_params_fixup = sof_pcm_dai_link_fixup; diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index a0cde053b61a..c410822d9920 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -12,6 +12,42 @@ #include "sof-priv.h" #include "sof-audio.h" +/* + * Helper function to determine the target DSP state during + * system suspend. This function only cares about the device + * D-states. Platform-specific substates, if any, should be + * handled by the platform-specific parts. + */ +static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev) +{ + u32 target_dsp_state; + + switch (sdev->system_suspend_target) { + case SOF_SUSPEND_S3: + /* DSP should be in D3 if the system is suspending to S3 */ + target_dsp_state = SOF_DSP_PM_D3; + break; + case SOF_SUSPEND_S0IX: + /* + * Currently, the only criterion for retaining the DSP in D0 + * is that there are streams that ignored the suspend trigger. + * Additional criteria such Soundwire clock-stop mode and + * device suspend latency considerations will be added later. + */ + if (snd_sof_stream_suspend_ignored(sdev)) + target_dsp_state = SOF_DSP_PM_D0; + else + target_dsp_state = SOF_DSP_PM_D3; + break; + default: + /* This case would be during runtime suspend */ + target_dsp_state = SOF_DSP_PM_D3; + break; + } + + return target_dsp_state; +} + static int sof_send_pm_ctx_ipc(struct snd_sof_dev *sdev, int cmd) { struct sof_ipc_pm_ctx pm_ctx; @@ -50,6 +86,7 @@ static void sof_cache_debugfs(struct snd_sof_dev *sdev) static int sof_resume(struct device *dev, bool runtime_resume) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + u32 old_state = sdev->dsp_power_state.state; int ret; /* do nothing if dsp resume callbacks are not set */ @@ -74,6 +111,10 @@ static int sof_resume(struct device *dev, bool runtime_resume) return ret; } + /* Nothing further to do if resuming from a low-power D0 substate */ + if (!runtime_resume && old_state == SOF_DSP_PM_D0) + return 0; + sdev->fw_state = SOF_FW_BOOT_PREPARE; /* load the firmware */ @@ -124,15 +165,13 @@ static int sof_resume(struct device *dev, bool runtime_resume) "error: ctx_restore ipc error during resume %d\n", ret); - /* initialize default D0 sub-state */ - sdev->d0_substate = SOF_DSP_D0I0; - return ret; } static int sof_suspend(struct device *dev, bool runtime_suspend) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + u32 target_state = 0; int ret; /* do nothing if dsp suspend callback is not set */ @@ -140,10 +179,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) return 0; if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) - goto power_down; - - /* release trace */ - snd_sof_release_trace(sdev); + goto suspend; /* set restore_stream for all streams during system suspend */ if (!runtime_suspend) { @@ -156,6 +192,15 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) } } + target_state = snd_sof_dsp_power_target(sdev); + + /* Skip to platform-specific suspend if DSP is entering D0 */ + if (target_state == SOF_DSP_PM_D0) + goto suspend; + + /* release trace */ + snd_sof_release_trace(sdev); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) /* cache debugfs contents during runtime suspend */ if (runtime_suspend) @@ -179,22 +224,26 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) ret); } -power_down: +suspend: /* return if the DSP was not probed successfully */ if (sdev->fw_state == SOF_FW_BOOT_NOT_STARTED) return 0; - /* power down all DSP cores */ + /* platform-specific suspend */ if (runtime_suspend) ret = snd_sof_dsp_runtime_suspend(sdev); else - ret = snd_sof_dsp_suspend(sdev); + ret = snd_sof_dsp_suspend(sdev, target_state); if (ret < 0) dev_err(sdev->dev, "error: failed to power down DSP during suspend %d\n", ret); + /* Do not reset FW state if DSP is in D0 */ + if (target_state == SOF_DSP_PM_D0) + return ret; + /* reset FW state */ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; @@ -221,112 +270,14 @@ int snd_sof_runtime_resume(struct device *dev) } EXPORT_SYMBOL(snd_sof_runtime_resume); -int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate) -{ - int ret; - - if (sdev->d0_substate == d0_substate) - return 0; - - /* do platform specific set_state */ - ret = snd_sof_dsp_set_power_state(sdev, d0_substate); - if (ret < 0) - return ret; - - /* update dsp D0 sub-state */ - sdev->d0_substate = d0_substate; - - return 0; -} -EXPORT_SYMBOL(snd_sof_set_d0_substate); - -/* - * Audio DSP states may transform as below:- - * - * D0I3 compatible stream - * Runtime +---------------------+ opened only, timeout - * suspend | +--------------------+ - * +------------+ D0(active) | | - * | | <---------------+ | - * | +--------> | | | - * | |Runtime +--^--+---------^--+--+ The last | | - * | |resume | | | | opened D0I3 | | - * | | | | | | compatible | | - * | | resume| | | | stream closed | | - * | | from | | D3 | | | | - * | | D3 | |suspend | | d0i3 | | - * | | | | | |suspend | | - * | | | | | | | | - * | | | | | | | | - * +-v---+-----------+--v-------+ | | +------+----v----+ - * | | | +-----------> | - * | D3 (suspended) | | | D0I3 +-----+ - * | | +--------------+ | | - * | | resume from | | | - * +-------------------^--------+ d0i3 suspend +----------------+ | - * | | - * | D3 suspend | - * +------------------------------------------------+ - * - * d0i3_suspend = s0_suspend && D0I3 stream opened, - * D3 suspend = !d0i3_suspend, - */ - int snd_sof_resume(struct device *dev) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); - int ret; - - if (snd_sof_dsp_d0i3_on_suspend(sdev)) { - /* resume from D0I3 */ - dev_dbg(sdev->dev, "DSP will exit from D0i3...\n"); - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0); - if (ret == -ENOTSUPP) { - /* fallback to resume from D3 */ - dev_dbg(sdev->dev, "D0i3 not supported, fall back to resume from D3...\n"); - goto d3_resume; - } else if (ret < 0) { - dev_err(sdev->dev, "error: failed to exit from D0I3 %d\n", - ret); - return ret; - } - - /* platform-specific resume from D0i3 */ - return snd_sof_dsp_resume(sdev); - } - -d3_resume: - /* resume from D3 */ return sof_resume(dev, false); } EXPORT_SYMBOL(snd_sof_resume); int snd_sof_suspend(struct device *dev) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); - int ret; - - if (snd_sof_dsp_d0i3_on_suspend(sdev)) { - /* suspend to D0i3 */ - dev_dbg(sdev->dev, "DSP is trying to enter D0i3...\n"); - ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); - if (ret == -ENOTSUPP) { - /* fallback to D3 suspend */ - dev_dbg(sdev->dev, "D0i3 not supported, fall back to D3...\n"); - goto d3_suspend; - } else if (ret < 0) { - dev_err(sdev->dev, "error: failed to enter D0I3, %d\n", - ret); - return ret; - } - - /* platform-specific suspend to D0i3 */ - return snd_sof_dsp_suspend(sdev); - } - -d3_suspend: - /* suspend to D3 */ return sof_suspend(dev, false); } EXPORT_SYMBOL(snd_sof_suspend); @@ -336,10 +287,13 @@ int snd_sof_prepare(struct device *dev) struct snd_sof_dev *sdev = dev_get_drvdata(dev); #if defined(CONFIG_ACPI) - sdev->s0_suspend = acpi_target_system_state() == ACPI_STATE_S0; + if (acpi_target_system_state() == ACPI_STATE_S0) + sdev->system_suspend_target = SOF_SUSPEND_S0IX; + else + sdev->system_suspend_target = SOF_SUSPEND_S3; #else /* will suspend to S3 by default */ - sdev->s0_suspend = false; + sdev->system_suspend_target = SOF_SUSPEND_S3; #endif return 0; @@ -350,6 +304,6 @@ void snd_sof_complete(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - sdev->s0_suspend = false; + sdev->system_suspend_target = SOF_SUSPEND_NONE; } EXPORT_SYMBOL(snd_sof_complete); diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c new file mode 100644 index 000000000000..c38169fe00c5 --- /dev/null +++ b/sound/soc/sof/probe.c @@ -0,0 +1,290 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski@intel.com> +// + +#include "sof-priv.h" +#include "probe.h" + +/** + * sof_ipc_probe_init - initialize data probing + * @sdev: SOF sound device + * @stream_tag: Extractor stream tag + * @buffer_size: DMA buffer size to set for extractor + * + * Host chooses whether extraction is supported or not by providing + * valid stream tag to DSP. Once specified, stream described by that + * tag will be tied to DSP for extraction for the entire lifetime of + * probe. + * + * Probing is initialized only once and each INIT request must be + * matched by DEINIT call. + */ +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, 1); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT; + msg->num_elems = 1; + msg->dma[0].stream_tag = stream_tag; + msg->dma[0].dma_buffer_size = buffer_size; + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_init); + +/** + * sof_ipc_probe_deinit - cleanup after data probing + * @sdev: SOF sound device + * + * Host sends DEINIT request to free previously initialized probe + * on DSP side once it is no longer needed. DEINIT only when there + * are no probes connected and with all injectors detached. + */ +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr msg; + struct sof_ipc_reply reply; + + msg.size = sizeof(msg); + msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT; + + return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size, + &reply, sizeof(reply)); +} +EXPORT_SYMBOL(sof_ipc_probe_deinit); + +static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, + void **params, size_t *num_params) +{ + struct sof_ipc_probe_info_params msg = {{{0}}}; + struct sof_ipc_probe_info_params *reply; + size_t bytes; + int ret; + + *params = NULL; + *num_params = 0; + + reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!reply) + return -ENOMEM; + msg.rhdr.hdr.size = sizeof(msg); + msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd; + + ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg, + msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE); + if (ret < 0 || reply->rhdr.error < 0) + goto exit; + + if (!reply->num_elems) + goto exit; + + if (cmd == SOF_IPC_PROBE_DMA_INFO) + bytes = sizeof(reply->dma[0]); + else + bytes = sizeof(reply->desc[0]); + bytes *= reply->num_elems; + *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); + if (!*params) { + ret = -ENOMEM; + goto exit; + } + *num_params = reply->num_elems; + +exit: + kfree(reply); + return ret; +} + +/** + * sof_ipc_probe_dma_info - retrieve list of active injection dmas + * @sdev: SOF sound device + * @dma: Returned list of active dmas + * @num_dma: Returned count of active dmas + * + * Host sends DMA_INFO request to obtain list of injection dmas it + * can use to transfer data over with. + * + * Note that list contains only injection dmas as there is only one + * extractor (dma) and it is always assigned on probing init. + * DSP knows exactly where data from extraction probes is going to, + * which is not the case for injection where multiple streams + * could be engaged. + */ +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO, + (void **)dma, num_dma); +} +EXPORT_SYMBOL(sof_ipc_probe_dma_info); + +/** + * sof_ipc_probe_dma_add - attach to specified dmas + * @sdev: SOF sound device + * @dma: List of streams (dmas) to attach to + * @num_dma: Number of elements in @dma + * + * Contrary to extraction, injection streams are never assigned + * on init. Before attempting any data injection, host is responsible + * for specifying streams which will be later used to transfer data + * to connected probe points. + */ +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, num_dma); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_dma; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD; + memcpy(&msg->dma[0], dma, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_add); + +/** + * sof_ipc_probe_dma_remove - detach from specified dmas + * @sdev: SOF sound device + * @stream_tag: List of stream tags to detach from + * @num_stream_tag: Number of elements in @stream_tag + * + * Host sends DMA_REMOVE request to free previously attached stream + * from being occupied for injection. Each detach operation should + * match equivalent DMA_ADD. Detach only when all probes tied to + * given stream have been disconnected. + */ +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag) +{ + struct sof_ipc_probe_dma_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, stream_tag, num_stream_tag); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_stream_tag; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE; + memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_remove); + +/** + * sof_ipc_probe_points_info - retrieve list of active probe points + * @sdev: SOF sound device + * @desc: Returned list of active probes + * @num_desc: Returned count of active probes + * + * Host sends PROBE_POINT_INFO request to obtain list of active probe + * points, valid for disconnection when given probe is no longer + * required. + */ +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO, + (void **)desc, num_desc); +} +EXPORT_SYMBOL(sof_ipc_probe_points_info); + +/** + * sof_ipc_probe_points_add - connect specified probes + * @sdev: SOF sound device + * @desc: List of probe points to connect + * @num_desc: Number of elements in @desc + * + * Dynamically connects to provided set of endpoints. Immediately + * after connection is established, host must be prepared to + * transfer data from or to target stream given the probing purpose. + * + * Each probe point should be removed using PROBE_POINT_REMOVE + * request when no longer needed. + */ +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc) +{ + struct sof_ipc_probe_point_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, desc, num_desc); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_desc; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD; + memcpy(&msg->desc[0], desc, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_add); + +/** + * sof_ipc_probe_points_remove - disconnect specified probes + * @sdev: SOF sound device + * @buffer_id: List of probe points to disconnect + * @num_buffer_id: Number of elements in @desc + * + * Removes previously connected probes from list of active probe + * points and frees all resources on DSP side. + */ +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id) +{ + struct sof_ipc_probe_point_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, buffer_id, num_buffer_id); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_buffer_id; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE; + memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_remove); diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h new file mode 100644 index 000000000000..45daa5552834 --- /dev/null +++ b/sound/soc/sof/probe.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski@intel.com> + */ + +#ifndef __SOF_PROBE_H +#define __SOF_PROBE_H + +#include <sound/sof/header.h> + +struct snd_sof_dev; + +#define SOF_PROBE_INVALID_NODE_ID UINT_MAX + +struct sof_probe_dma { + unsigned int stream_tag; + unsigned int dma_buffer_size; +} __packed; + +enum sof_connection_purpose { + SOF_CONNECTION_PURPOSE_EXTRACT = 1, + SOF_CONNECTION_PURPOSE_INJECT, +}; + +struct sof_probe_point_desc { + unsigned int buffer_id; + unsigned int purpose; + unsigned int stream_tag; +} __packed; + +struct sof_ipc_probe_dma_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_dma dma[0]; +} __packed; + +struct sof_ipc_probe_info_params { + struct sof_ipc_reply rhdr; + unsigned int num_elems; + union { + struct sof_probe_dma dma[0]; + struct sof_probe_point_desc desc[0]; + }; +} __packed; + +struct sof_ipc_probe_dma_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int stream_tag[0]; +} __packed; + +struct sof_ipc_probe_point_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_point_desc desc[0]; +} __packed; + +struct sof_ipc_probe_point_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int buffer_id[0]; +} __packed; + +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size); +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev); +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma); +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma); +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag); +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc); +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc); +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id); + +#endif diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0d8f65b9ae25..fc4ed2a8a914 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -11,7 +11,40 @@ #include "sof-audio.h" #include "ops.h" -bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev) +/* + * helper to determine if there are only D0i3 compatible + * streams active + */ +bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) +{ + struct snd_pcm_substream *substream; + struct snd_sof_pcm *spcm; + bool d0i3_compatible_active = false; + int dir; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + for_each_pcm_streams(dir) { + substream = spcm->stream[dir].substream; + if (!substream || !substream->runtime) + continue; + + /* + * substream->runtime being not NULL indicates that + * that the stream is open. No need to check the + * stream state. + */ + if (!spcm->stream[dir].d0i3_compatible) + return false; + + d0i3_compatible_active = true; + } + } + + return d0i3_compatible_active; +} +EXPORT_SYMBOL(snd_sof_dsp_only_d0i3_compatible_stream_active); + +bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev) { struct snd_sof_pcm *spcm; @@ -38,7 +71,14 @@ int sof_set_hw_params_upon_resume(struct device *dev) * have been suspended. */ list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { + /* + * do not reset hw_params upon resume for streams that + * were kept running during suspend + */ + if (spcm->stream[dir].suspend_ignored) + continue; + substream = spcm->stream[dir].substream; if (!substream || !substream->runtime) continue; @@ -279,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - dir = SNDRV_PCM_STREAM_PLAYBACK; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; - } - - dir = SNDRV_PCM_STREAM_CAPTURE; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; + for_each_pcm_streams(dir) { + if (spcm->stream[dir].comp_id == comp_id) { + *direction = dir; + return spcm; + } } } diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index a62fb2da6a6e..bf65f31af858 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -11,6 +11,8 @@ #ifndef __SOUND_SOC_SOF_AUDIO_H #define __SOUND_SOC_SOF_AUDIO_H +#include <linux/workqueue.h> + #include <sound/soc.h> #include <sound/control.h> #include <sound/sof/stream.h> /* needs to be included before control.h */ @@ -189,6 +191,7 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_soc_component *scomp, unsigned int pcm_id); void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream); +void snd_sof_pcm_period_elapsed_work(struct work_struct *work); /* * Mixer IPC @@ -202,7 +205,8 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, /* PM */ int sof_restore_pipelines(struct device *dev); int sof_set_hw_params_upon_resume(struct device *dev); -bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev); +bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev); +bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev); /* Machine driver enumeration */ int sof_machine_register(struct snd_sof_dev *sdev, void *pdata); diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 39ea8af6213f..16e49f2ee629 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -13,12 +13,21 @@ #include "ops.h" extern struct snd_sof_dsp_ops sof_imx8_ops; +extern struct snd_sof_dsp_ops sof_imx8x_ops; /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) static struct sof_dev_desc sof_of_imx8qxp_desc = { .default_fw_path = "imx/sof", .default_tplg_path = "imx/sof-tplg", + .default_fw_filename = "sof-imx8x.ri", + .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", + .ops = &sof_imx8x_ops, +}; + +static struct sof_dev_desc sof_of_imx8qm_desc = { + .default_fw_path = "imx/sof", + .default_tplg_path = "imx/sof-tplg", .default_fw_filename = "sof-imx8.ri", .nocodec_tplg_filename = "sof-imx8-nocodec.tplg", .ops = &sof_imx8_ops, @@ -103,6 +112,7 @@ static int sof_of_remove(struct platform_device *pdev) static const struct of_device_id sof_of_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) { .compatible = "fsl,imx8qxp-dsp", .data = &sof_of_imx8qxp_desc}, + { .compatible = "fsl,imx8qm-dsp", .data = &sof_of_imx8qm_desc}, #endif { } }; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index bc2337cf1142..a4b297c842df 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -54,10 +54,26 @@ extern int sof_core_debug; (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) || \ IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)) -/* DSP D0ix sub-state */ -enum sof_d0_substate { - SOF_DSP_D0I0 = 0, /* DSP default D0 substate */ - SOF_DSP_D0I3, /* DSP D0i3(low power) substate*/ +/* DSP power state */ +enum sof_dsp_power_states { + SOF_DSP_PM_D0, + SOF_DSP_PM_D1, + SOF_DSP_PM_D2, + SOF_DSP_PM_D3_HOT, + SOF_DSP_PM_D3, + SOF_DSP_PM_D3_COLD, +}; + +struct sof_dsp_power_state { + u32 state; + u32 substate; /* platform-specific */ +}; + +/* System suspend target state */ +enum sof_system_suspend_state { + SOF_SUSPEND_NONE = 0, + SOF_SUSPEND_S0IX, + SOF_SUSPEND_S3, }; struct snd_sof_dev; @@ -154,6 +170,27 @@ struct snd_sof_dsp_ops { snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* Except for probe_pointer, all probe ops are mandatory */ + int (*probe_assign)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_free)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_set_params)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_trigger)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_pointer)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); /* optional */ +#endif + /* host read DSP stream data */ void (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, @@ -169,14 +206,15 @@ struct snd_sof_dsp_ops { int (*post_fw_run)(struct snd_sof_dev *sof_dev); /* optional */ /* DSP PM */ - int (*suspend)(struct snd_sof_dev *sof_dev); /* optional */ + int (*suspend)(struct snd_sof_dev *sof_dev, + u32 target_state); /* optional */ int (*resume)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_suspend)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_resume)(struct snd_sof_dev *sof_dev); /* optional */ int (*runtime_idle)(struct snd_sof_dev *sof_dev); /* optional */ int (*set_hw_params_upon_resume)(struct snd_sof_dev *sdev); /* optional */ int (*set_power_state)(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); /* optional */ + const struct sof_dsp_power_state *target_state); /* optional */ /* DSP clocking */ int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); /* optional */ @@ -323,10 +361,11 @@ struct snd_sof_dev { */ struct snd_soc_component_driver plat_drv; - /* power states related */ - enum sof_d0_substate d0_substate; - /* flag to track if the intended power target of suspend is S0ix */ - bool s0_suspend; + /* current DSP power state */ + struct sof_dsp_power_state dsp_power_state; + + /* Intended power target of system suspend */ + enum sof_system_suspend_state system_suspend_target; /* DSP firmware boot */ wait_queue_head_t boot_wait; @@ -376,16 +415,15 @@ struct snd_sof_dev { u32 enabled_cores_mask; /* keep track of enabled cores */ /* FW configuration */ - struct sof_ipc_dma_buffer_data *info_buffer; struct sof_ipc_window *info_window; /* IPC timeouts in ms */ int ipc_timeout; int boot_timeout; - /* Wait queue for code loading */ - wait_queue_head_t waitq; - int code_loading; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + unsigned int extractor_stream_tag; +#endif /* DMA for Trace */ struct snd_dma_buffer dmatb; @@ -417,8 +455,6 @@ int snd_sof_resume(struct device *dev); int snd_sof_suspend(struct device *dev); int snd_sof_prepare(struct device *dev); void snd_sof_complete(struct device *dev); -int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, - enum sof_d0_substate d0_substate); void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); @@ -454,6 +490,9 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev); int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, void *msg_data, size_t msg_bytes, void *reply_data, size_t reply_bytes); +int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header, + void *msg_data, size_t msg_bytes, + void *reply_data, size_t reply_bytes); /* * Trace/debug diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9f4f8868b386..fe8ba3e05e08 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -9,6 +9,7 @@ // #include <linux/firmware.h> +#include <linux/workqueue.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <uapi/sound/sof/tokens.h> @@ -1240,6 +1241,8 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, { struct snd_soc_card *card = scomp->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *cpu_dai; + int i; list_for_each_entry(rtd, &card->rtd_list, list) { dev_vdbg(scomp->dev, "tplg: check widget: %s stream: %s dai stream: %s\n", @@ -1254,13 +1257,15 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, switch (w->id) { case snd_soc_dapm_dai_out: - rtd->cpu_dai->capture_widget = w; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + cpu_dai->capture_widget = w; dai->name = rtd->dai_link->name; dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); break; case snd_soc_dapm_dai_in: - rtd->cpu_dai->playback_widget = w; + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + cpu_dai->playback_widget = w; dai->name = rtd->dai_link->name; dev_dbg(scomp->dev, "tplg: connected widget %s -> DAI link %s\n", w->name, rtd->dai_link->name); @@ -2444,7 +2449,7 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_stream_caps *caps; struct snd_soc_tplg_private *private = &pcm->priv; struct snd_sof_pcm *spcm; - int stream = SNDRV_PCM_STREAM_PLAYBACK; + int stream; int ret = 0; /* nothing to do for BEs atm */ @@ -2456,8 +2461,12 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, return -ENOMEM; spcm->scomp = scomp; - spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id = COMP_ID_UNASSIGNED; - spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id = COMP_ID_UNASSIGNED; + + for_each_pcm_streams(stream) { + spcm->stream[stream].comp_id = COMP_ID_UNASSIGNED; + INIT_WORK(&spcm->stream[stream].period_elapsed_work, + snd_sof_pcm_period_elapsed_work); + } spcm->pcm = *pcm; dev_dbg(scomp->dev, "tplg: load pcm %s\n", pcm->dai_name); @@ -2478,8 +2487,10 @@ static int sof_dai_load(struct snd_soc_component *scomp, int index, if (!spcm->pcm.playback) goto capture; + stream = SNDRV_PCM_STREAM_PLAYBACK; + dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: playback d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[0].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; @@ -2509,7 +2520,7 @@ capture: return ret; dev_vdbg(scomp->dev, "tplg: pcm %s stream tokens: capture d0i3:%d\n", - spcm->pcm.pcm_name, spcm->stream[1].d0i3_compatible); + spcm->pcm.pcm_name, spcm->stream[stream].d0i3_compatible); caps = &spcm->pcm.caps[stream]; diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 5474fd3de8c0..5e0ac8278572 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -8,7 +8,7 @@ config SND_SOC_SPRD the Spreadtrum SoCs' Audio interfaces. config SND_SOC_SPRD_MCDT - bool "Spreadtrum multi-channel data transfer support" + tristate "Spreadtrum multi-channel data transfer support" depends on SND_SOC_SPRD help Say y here to enable multi-channel data transfer support. It diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h index 9cc7e207ac76..679e3af3baad 100644 --- a/sound/soc/sprd/sprd-mcdt.h +++ b/sound/soc/sprd/sprd-mcdt.h @@ -48,7 +48,7 @@ struct sprd_mcdt_chan { struct list_head list; }; -#ifdef CONFIG_SND_SOC_SPRD_MCDT +#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT) struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type); void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index 6cddf551bc11..74d48340cade 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -135,7 +135,7 @@ static int sprd_platform_compr_dma_config(struct snd_compr_stream *cstream, struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct device *dev = component->dev; - struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct sprd_pcm_dma_params *dma_params = data->dma_params; struct sprd_compr_dma *dma = &stream->dma[channel]; struct dma_slave_config config = { }; @@ -321,7 +321,7 @@ static int sprd_platform_compr_open(struct snd_compr_stream *cstream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct device *dev = component->dev; - struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct sprd_compr_stream *stream; struct sprd_compr_callback cb; int stream_id = cstream->direction, ret; diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index 2284558684bc..d12d3cad8cbd 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -200,7 +200,7 @@ static int sprd_pcm_hw_params(struct snd_soc_component *component, unsigned long flags; int ret, i, j, sg_num; - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_params = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); if (!dma_params) { dev_warn(component->dev, "no dma parameters setting\n"); dma_private->params = NULL; diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 51407a21c440..16ff02953015 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -215,7 +215,7 @@ static int stm32_adfsdm_trigger(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -235,7 +235,7 @@ static int stm32_adfsdm_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct stm32_adfsdm_priv *priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); int ret; ret = snd_soc_set_runtime_hwparams(substream, &stm32_adfsdm_pcm_hw); @@ -250,7 +250,7 @@ static int stm32_adfsdm_pcm_close(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); priv->substream = NULL; @@ -263,7 +263,7 @@ static snd_pcm_uframes_t stm32_adfsdm_pcm_pointer( { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); return bytes_to_frames(substream->runtime, priv->pos); } @@ -274,7 +274,7 @@ static int stm32_adfsdm_pcm_hw_params(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); priv->pcm_buff = substream->runtime->dma_area; @@ -287,7 +287,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, { struct snd_pcm *pcm = rtd->pcm; struct stm32_adfsdm_priv *priv = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); unsigned int size = DFSDM_MAX_PERIODS * DFSDM_MAX_PERIOD_SIZE; snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 3e7226a53e53..7c4d63c33f15 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -831,25 +831,33 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, /* Get clocks */ i2s->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(i2s->pclk)) { - dev_err(&pdev->dev, "Could not get pclk\n"); + if (PTR_ERR(i2s->pclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get pclk: %ld\n", + PTR_ERR(i2s->pclk)); return PTR_ERR(i2s->pclk); } i2s->i2sclk = devm_clk_get(&pdev->dev, "i2sclk"); if (IS_ERR(i2s->i2sclk)) { - dev_err(&pdev->dev, "Could not get i2sclk\n"); + if (PTR_ERR(i2s->i2sclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get i2sclk: %ld\n", + PTR_ERR(i2s->i2sclk)); return PTR_ERR(i2s->i2sclk); } i2s->x8kclk = devm_clk_get(&pdev->dev, "x8k"); if (IS_ERR(i2s->x8kclk)) { - dev_err(&pdev->dev, "missing x8k parent clock\n"); + if (PTR_ERR(i2s->x8kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get x8k parent clock: %ld\n", + PTR_ERR(i2s->x8kclk)); return PTR_ERR(i2s->x8kclk); } i2s->x11kclk = devm_clk_get(&pdev->dev, "x11k"); if (IS_ERR(i2s->x11kclk)) { - dev_err(&pdev->dev, "missing x11k parent clock\n"); + if (PTR_ERR(i2s->x11kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get x11k parent clock: %ld\n", + PTR_ERR(i2s->x11kclk)); return PTR_ERR(i2s->x11kclk); } @@ -866,12 +874,24 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, } /* Reset */ - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); + + return 0; +} + +static int stm32_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } @@ -903,42 +923,51 @@ static int stm32_i2s_probe(struct platform_device *pdev) i2s->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "pclk", i2s->base, i2s->regmap_conf); if (IS_ERR(i2s->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); + if (PTR_ERR(i2s->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(i2s->regmap)); return PTR_ERR(i2s->regmap); } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_i2s_component, - i2s->dai_drv, 1); - if (ret) + ret = snd_dmaengine_pcm_register(&pdev->dev, &stm32_i2s_pcm_config, 0); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; + } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, - &stm32_i2s_pcm_config, 0); - if (ret) + ret = snd_soc_register_component(&pdev->dev, &stm32_i2s_component, + i2s->dai_drv, 1); + if (ret) { + snd_dmaengine_pcm_unregister(&pdev->dev); return ret; + } /* Set SPI/I2S in i2s mode */ ret = regmap_update_bits(i2s->regmap, STM32_I2S_CGFR_REG, I2S_CGFR_I2SMOD, I2S_CGFR_I2SMOD); if (ret) - return ret; + goto error; ret = regmap_read(i2s->regmap, STM32_I2S_IPIDR_REG, &val); if (ret) - return ret; + goto error; if (val == I2S_IPIDR_NUMBER) { ret = regmap_read(i2s->regmap, STM32_I2S_HWCFGR_REG, &val); if (ret) - return ret; + goto error; if (!FIELD_GET(I2S_HWCFGR_I2S_SUPPORT_MASK, val)) { dev_err(&pdev->dev, "Device does not support i2s mode\n"); - return -EPERM; + ret = -EPERM; + goto error; } ret = regmap_read(i2s->regmap, STM32_I2S_VERR_REG, &val); + if (ret) + goto error; dev_dbg(&pdev->dev, "I2S version: %lu.%lu registered\n", FIELD_GET(I2S_VERR_MAJ_MASK, val), @@ -946,6 +975,11 @@ static int stm32_i2s_probe(struct platform_device *pdev) } return ret; + +error: + stm32_i2s_remove(pdev); + + return ret; } MODULE_DEVICE_TABLE(of, stm32_i2s_ids); @@ -981,6 +1015,7 @@ static struct platform_driver stm32_i2s_driver = { .pm = &stm32_i2s_pm_ops, }, .probe = stm32_i2s_probe, + .remove = stm32_i2s_remove, }; module_platform_driver(stm32_i2s_driver); diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index e20267504b16..058757c721f0 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -174,20 +174,26 @@ static int stm32_sai_probe(struct platform_device *pdev) if (!STM_SAI_IS_F4(sai)) { sai->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(sai->pclk)) { - dev_err(&pdev->dev, "missing bus clock pclk\n"); + if (PTR_ERR(sai->pclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing bus clock pclk: %ld\n", + PTR_ERR(sai->pclk)); return PTR_ERR(sai->pclk); } } sai->clk_x8k = devm_clk_get(&pdev->dev, "x8k"); if (IS_ERR(sai->clk_x8k)) { - dev_err(&pdev->dev, "missing x8k parent clock\n"); + if (PTR_ERR(sai->clk_x8k) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing x8k parent clock: %ld\n", + PTR_ERR(sai->clk_x8k)); return PTR_ERR(sai->clk_x8k); } sai->clk_x11k = devm_clk_get(&pdev->dev, "x11k"); if (IS_ERR(sai->clk_x11k)) { - dev_err(&pdev->dev, "missing x11k parent clock\n"); + if (PTR_ERR(sai->clk_x11k) != -EPROBE_DEFER) + dev_err(&pdev->dev, "missing x11k parent clock: %ld\n", + PTR_ERR(sai->clk_x11k)); return PTR_ERR(sai->clk_x11k); } @@ -197,12 +203,16 @@ static int stm32_sai_probe(struct platform_device *pdev) return sai->irq; /* reset */ - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); /* Enable peripheral clock to allow register access */ ret = clk_prepare_enable(sai->pclk); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 10eb4b8e8e7e..2bd280c01c33 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1238,7 +1238,7 @@ static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct stm32_sai_sub_data *sai = dev_get_drvdata(cpu_dai->dev); int *ptr = (int *)(runtime->dma_area + hwoff + channel * (runtime->dma_bytes / runtime->channels)); @@ -1380,7 +1380,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, sai->regmap_config); if (IS_ERR(sai->regmap)) { - dev_err(&pdev->dev, "Failed to initialize MMIO\n"); + if (PTR_ERR(sai->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(sai->regmap)); return PTR_ERR(sai->regmap); } @@ -1471,7 +1473,9 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, of_node_put(args.np); sai->sai_ck = devm_clk_get(&pdev->dev, "sai_ck"); if (IS_ERR(sai->sai_ck)) { - dev_err(&pdev->dev, "Missing kernel clock sai_ck\n"); + if (PTR_ERR(sai->sai_ck) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Missing kernel clock sai_ck: %ld\n", + PTR_ERR(sai->sai_ck)); return PTR_ERR(sai->sai_ck); } @@ -1545,7 +1549,8 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not register pcm dma\n"); return ret; } diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbe..1bfa3b2ba974 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -406,7 +406,9 @@ static int stm32_spdifrx_dma_ctrl_register(struct device *dev, spdifrx->ctrl_chan = dma_request_chan(dev, "rx-ctrl"); if (IS_ERR(spdifrx->ctrl_chan)) { - dev_err(dev, "dma_request_slave_channel failed\n"); + if (PTR_ERR(spdifrx->ctrl_chan) != -EPROBE_DEFER) + dev_err(dev, "dma_request_slave_channel error %ld\n", + PTR_ERR(spdifrx->ctrl_chan)); return PTR_ERR(spdifrx->ctrl_chan); } @@ -929,7 +931,9 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev, spdifrx->kclk = devm_clk_get(&pdev->dev, "kclk"); if (IS_ERR(spdifrx->kclk)) { - dev_err(&pdev->dev, "Could not get kclk\n"); + if (PTR_ERR(spdifrx->kclk) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Could not get kclk: %ld\n", + PTR_ERR(spdifrx->kclk)); return PTR_ERR(spdifrx->kclk); } @@ -940,6 +944,22 @@ static int stm32_spdifrx_parse_of(struct platform_device *pdev, return 0; } +static int stm32_spdifrx_remove(struct platform_device *pdev) +{ + struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); + + if (spdifrx->ctrl_chan) + dma_release_channel(spdifrx->ctrl_chan); + + if (spdifrx->dmab) + snd_dma_free_pages(spdifrx->dmab); + + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + + return 0; +} + static int stm32_spdifrx_probe(struct platform_device *pdev) { struct stm32_spdifrx_data *spdifrx; @@ -967,7 +987,9 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) spdifrx->base, spdifrx->regmap_conf); if (IS_ERR(spdifrx->regmap)) { - dev_err(&pdev->dev, "Regmap init failed\n"); + if (PTR_ERR(spdifrx->regmap) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Regmap init error %ld\n", + PTR_ERR(spdifrx->regmap)); return PTR_ERR(spdifrx->regmap); } @@ -978,37 +1000,46 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) return ret; } - rst = devm_reset_control_get_exclusive(&pdev->dev, NULL); - if (!IS_ERR(rst)) { - reset_control_assert(rst); - udelay(2); - reset_control_deassert(rst); + rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(rst)) { + if (PTR_ERR(rst) != -EPROBE_DEFER) + dev_err(&pdev->dev, "Reset controller error %ld\n", + PTR_ERR(rst)); + return PTR_ERR(rst); } + reset_control_assert(rst); + udelay(2); + reset_control_deassert(rst); - ret = devm_snd_soc_register_component(&pdev->dev, - &stm32_spdifrx_component, - stm32_spdifrx_dai, - ARRAY_SIZE(stm32_spdifrx_dai)); - if (ret) + pcm_config = &stm32_spdifrx_pcm_config; + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; + } + + ret = snd_soc_register_component(&pdev->dev, + &stm32_spdifrx_component, + stm32_spdifrx_dai, + ARRAY_SIZE(stm32_spdifrx_dai)); + if (ret) { + snd_dmaengine_pcm_unregister(&pdev->dev); + return ret; + } ret = stm32_spdifrx_dma_ctrl_register(&pdev->dev, spdifrx); if (ret) goto error; - pcm_config = &stm32_spdifrx_pcm_config; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, pcm_config, 0); - if (ret) { - dev_err(&pdev->dev, "PCM DMA register returned %d\n", ret); - goto error; - } - ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_IDR, &idr); if (ret) goto error; if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), @@ -1018,27 +1049,11 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) return ret; error: - if (!IS_ERR(spdifrx->ctrl_chan)) - dma_release_channel(spdifrx->ctrl_chan); - if (spdifrx->dmab) - snd_dma_free_pages(spdifrx->dmab); + stm32_spdifrx_remove(pdev); return ret; } -static int stm32_spdifrx_remove(struct platform_device *pdev) -{ - struct stm32_spdifrx_data *spdifrx = platform_get_drvdata(pdev); - - if (spdifrx->ctrl_chan) - dma_release_channel(spdifrx->ctrl_chan); - - if (spdifrx->dmab) - snd_dma_free_pages(spdifrx->dmab); - - return 0; -} - MODULE_DEVICE_TABLE(of, stm32_spdifrx_ids); #ifdef CONFIG_PM_SLEEP diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index 98a9fe645521..86779a99df75 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -244,7 +244,7 @@ static int sun4i_spdif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -EINVAL; diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 686561df8e13..ca51af114419 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -86,7 +86,6 @@ #define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9) struct sun8i_codec { - struct device *dev; struct regmap *regmap; struct clk *clk_module; struct clk *clk_bus; @@ -542,8 +541,6 @@ static int sun8i_codec_probe(struct platform_device *pdev) if (!scodec) return -ENOMEM; - scodec->dev = &pdev->dev; - scodec->clk_module = devm_clk_get(&pdev->dev, "mod"); if (IS_ERR(scodec->clk_module)) { dev_err(&pdev->dev, "Failed to get the module clock\n"); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 9e8b1497efd3..ec39ecba1e8b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -37,7 +37,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 4954a33ff46b..d800b62b36f8 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -38,7 +38,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index d46915a3ec4c..9878bc3eb89e 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -40,7 +40,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 81cb6cc6236e..5821313db977 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -42,7 +42,7 @@ static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); int srate, mclk, err; diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index e13b81d29cf3..dc411ba2e36d 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -36,7 +36,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index f6dd790dad71..0d653a605358 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -40,7 +40,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index f08d3489c3cf..9b5651502f12 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -45,7 +45,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -143,19 +143,37 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, return 0; } +static int tegra_wm8903_event_int_mic(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_int_mic_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_int_mic_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk), SND_SOC_DAPM_HP("Headphone Jack", tegra_wm8903_event_hp), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", tegra_wm8903_event_int_mic), }; static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); @@ -187,7 +205,7 @@ static int tegra_wm8903_remove(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; wm8903_mic_detect(component, NULL, 0, 0); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 3f67ddd13674..f9834afaa2e8 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -35,7 +35,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 29f61053ab62..c5408c129f34 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only menu "Audio support for Texas Instruments SoCs" -depends on DMA_OMAP || TI_EDMA || COMPILE_TEST +depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST config SND_SOC_TI_EDMA_PCM tristate @@ -10,6 +10,10 @@ config SND_SOC_TI_SDMA_PCM tristate select SND_SOC_GENERIC_DMAENGINE_PCM +config SND_SOC_TI_UDMA_PCM + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + comment "Texas Instruments DAI support for:" config SND_SOC_DAVINCI_ASP tristate "daVinci Audio Serial Port (ASP) or McBSP support" @@ -24,6 +28,7 @@ config SND_SOC_DAVINCI_MCASP tristate "Multichannel Audio Serial Port (McASP) support" select SND_SOC_TI_EDMA_PCM select SND_SOC_TI_SDMA_PCM + select SND_SOC_TI_UDMA_PCM help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -31,6 +36,7 @@ config SND_SOC_DAVINCI_MCASP - Sitara line of SoCs (AM335x, AM438x, etc) - DRA7x devices - Keystone devices + - K3 devices (am654, j721e) config SND_SOC_DAVINCI_VCIF tristate "daVinci Voice Interface (VCIF) support" diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile index 08c44d56ef3e..ea48c6679cc7 100644 --- a/sound/soc/ti/Makefile +++ b/sound/soc/ti/Makefile @@ -3,9 +3,11 @@ # Platform drivers snd-soc-ti-edma-objs := edma-pcm.o snd-soc-ti-sdma-objs := sdma-pcm.o +snd-soc-ti-udma-objs := udma-pcm.o obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o +obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o # CPU DAI drivers snd-soc-davinci-asp-objs := davinci-i2s.o diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 8e2fb81ad05c..e17cd5e939f0 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -460,14 +460,14 @@ static void ams_delta_shutdown(struct snd_pcm_substream *substream) static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct snd_soc_dapm_context *dapm = &card->dapm; int ret; /* Codec is ready, now add/activate board specific controls */ /* Store a pointer to the codec structure for tty ldisc use */ - cx20442_codec = rtd->codec_dai->component; + cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component; /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 686b23d7a90d..2cfbeebdfb41 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -54,8 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_card *soc_card = rtd->card; int ret = 0; unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index e1e937eb1dc1..734ffe925c4d 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -38,6 +38,7 @@ #include "edma-pcm.h" #include "sdma-pcm.h" +#include "udma-pcm.h" #include "davinci-mcasp.h" #define MCASP_MAX_AFIFO_DEPTH 64 @@ -1764,10 +1765,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( } else if (match) { pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - ret = -ENOMEM; - return pdata; - } + if (!pdata) + return NULL; } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; @@ -1875,6 +1874,7 @@ nodata: enum { PCM_EDMA, PCM_SDMA, + PCM_UDMA, }; static const char *sdma_prefix = "ti,omap"; @@ -1912,6 +1912,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) return PCM_SDMA; + else if (strstr(tmp, "udmap")) + return PCM_UDMA; return PCM_EDMA; } @@ -2371,6 +2373,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) case PCM_SDMA: ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); break; + case PCM_UDMA: + ret = udma_pcm_platform_register(&pdev->dev); + break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); case -EPROBE_DEFER: diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c index c84650e4a7aa..ee4d3ef821a1 100644 --- a/sound/soc/ti/davinci-vcif.c +++ b/sound/soc/ti/davinci-vcif.c @@ -43,7 +43,7 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; u32 w; @@ -62,7 +62,7 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_vcif_dev *davinci_vcif_dev = - snd_soc_dai_get_drvdata(rtd->cpu_dai); + snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; u32 w; diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c index 3ad2b6daf31e..a1672b479cb7 100644 --- a/sound/soc/ti/n810.c +++ b/sound/soc/ti/n810.c @@ -101,7 +101,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int err; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c index 6d564ab5e437..61e45fea5dd8 100644 --- a/sound/soc/ti/omap-abe-twl6040.c +++ b/sound/soc/ti/omap-abe-twl6040.c @@ -46,7 +46,7 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int clk_id, freq; @@ -78,7 +78,7 @@ static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret = 0; ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, @@ -166,7 +166,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_card *card = rtd->card; struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 1a3fe854e856..5a32b54bbf3b 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -489,7 +489,7 @@ OMAP_MCBSP_ST_CONTROLS(3); int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); if (!mcbsp->st_data) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 302d5c493c29..3d41ca2238d4 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -737,7 +737,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, unsigned int packet_size) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int words; @@ -902,7 +902,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); u16 fifo_use; snd_pcm_sframes_t delay; diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index d7ac4df6f2d9..f2dbadea33bb 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -532,7 +532,7 @@ static const struct snd_soc_component_driver omap_mcpdm_component = { void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd, u8 rx1, u8 rx2) { - struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2); } diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c index 545f8dac9bd5..b04146311b31 100644 --- a/sound/soc/ti/omap3pandora.c +++ b/sound/soc/ti/omap3pandora.c @@ -32,8 +32,8 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c index 1ca466bc4025..e01485cc51a1 100644 --- a/sound/soc/ti/osk5912.c +++ b/sound/soc/ti/osk5912.c @@ -39,7 +39,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int err; /* Set the codec system clock for DAC and ADC */ diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index fdb0dc85fe67..2a714a004163 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -103,7 +103,7 @@ static int rx51_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); /* Set the codec system clock for DAC and ADC */ return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000, diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c new file mode 100644 index 000000000000..39830caaaf7c --- /dev/null +++ b/sound/soc/ti/udma-pcm.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "udma-pcm.h" + +static const struct snd_pcm_hardware udma_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP | + SNDRV_PCM_INFO_INTERLEAVED, + .buffer_bytes_max = SIZE_MAX, + .period_bytes_min = 32, + .period_bytes_max = SZ_64K, + .periods_min = 2, + .periods_max = UINT_MAX, +}; + +static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = { + .pcm_hardware = &udma_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + +int udma_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config, + 0); +} +EXPORT_SYMBOL_GPL(udma_pcm_platform_register); + +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); +MODULE_DESCRIPTION("UDMA PCM ASoC platform driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h new file mode 100644 index 000000000000..54111e7312c1 --- /dev/null +++ b/sound/soc/ti/udma-pcm.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + */ + +#ifndef __UDMA_PCM_H__ +#define __UDMA_PCM_H__ + +#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM) +int udma_pcm_platform_register(struct device *dev); +#else +static inline int udma_pcm_platform_register(struct device *dev) +{ + return 0; +} +#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */ + +#endif /* __UDMA_PCM_H__ */ diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 985487cc3a55..4b1cd4da3e36 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -269,7 +269,7 @@ static int txx9aclc_pcm_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(component->dev); struct txx9aclc_soc_device *dev; diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 17f773ac5ca1..232d3cc5bce0 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -23,7 +23,7 @@ static int uniphier_aio_comprdma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_compr *compr = rtd->compr; struct device *dev = compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[compr->direction]; size_t size = AUD_RING_SIZE; int dma_dir = DMA_FROM_DEVICE, ret; @@ -56,7 +56,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd) { struct snd_compr *compr = rtd->compr; struct device *dev = compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[compr->direction]; int dma_dir = DMA_FROM_DEVICE; @@ -73,7 +73,7 @@ static int uniphier_aio_comprdma_free(struct snd_soc_pcm_runtime *rtd) static int uniphier_aio_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int ret; @@ -98,7 +98,7 @@ static int uniphier_aio_compr_open(struct snd_compr_stream *cstream) static int uniphier_aio_compr_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int ret; @@ -118,7 +118,7 @@ static int uniphier_aio_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; *params = sub->cparams.codec; @@ -130,7 +130,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; struct device *dev = &aio->chip->pdev->dev; int ret; @@ -165,7 +165,7 @@ static int uniphier_aio_compr_set_params(struct snd_compr_stream *cstream, static int uniphier_aio_compr_hw_free(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; sub->setting = 0; @@ -177,7 +177,7 @@ static int uniphier_aio_compr_prepare(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int bytes = runtime->fragment_size; unsigned long flags; @@ -215,7 +215,7 @@ static int uniphier_aio_compr_trigger(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; struct device *dev = &aio->chip->pdev->dev; int bytes = runtime->fragment_size, ret = 0; @@ -248,7 +248,7 @@ static int uniphier_aio_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; int bytes = runtime->fragment_size; unsigned long flags; @@ -322,7 +322,7 @@ static int uniphier_aio_compr_copy(struct snd_compr_stream *cstream, struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *runtime = cstream->runtime; struct device *carddev = rtd->compr->card->dev; - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[cstream->direction]; size_t cnt = min_t(size_t, count, aio_rb_space_to_end(sub) / 2); int bytes = runtime->fragment_size; diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index da83423c52e2..4bbcb007df41 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -109,7 +109,7 @@ static int uniphier_aiodma_prepare(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; int bytes = runtime->period_size * runtime->channels * samples_to_bytes(runtime, 1); @@ -136,7 +136,7 @@ static int uniphier_aiodma_trigger(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; struct device *dev = &aio->chip->pdev->dev; int bytes = runtime->period_size * @@ -172,7 +172,7 @@ static snd_pcm_uframes_t uniphier_aiodma_pointer( { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct uniphier_aio *aio = uniphier_priv(rtd->cpu_dai); + struct uniphier_aio *aio = uniphier_priv(asoc_rtd_to_cpu(rtd, 0)); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; int bytes = runtime->period_size * runtime->channels * samples_to_bytes(runtime, 1); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 77655084bbde..6aaa19829a73 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -215,8 +215,8 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct device *dev = rtd->card->dev; unsigned int fmt; int channels, ret = 0, driver_mode, slots; @@ -339,7 +339,7 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); mutex_lock(&mop500_ab8500_params_lock); __clear_bit(cpu_dai->id, &mop500_ab8500_usage); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 9445dbe8e039..39b96c132bc8 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -46,7 +46,7 @@ static const struct snd_pcm_hardware ux500_pcm_hw = { static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { - struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_soc_dai *dai = asoc_rtd_to_cpu(rtd, 0); u16 per_data_width, mem_data_width; struct stedma40_chan_cfg *dma_cfg; struct ux500_msp_dma_params *dma_params; @@ -86,7 +86,7 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct msp_i2s_platform_data *pdata = rtd->cpu_dai->dev->platform_data; + struct msp_i2s_platform_data *pdata = asoc_rtd_to_cpu(rtd, 0)->dev->platform_data; struct snd_dmaengine_dai_dma_data *snd_dma_params; struct ux500_msp_dma_params *ste_dma_params; dma_addr_t dma_addr; @@ -94,11 +94,11 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, if (pdata) { ste_dma_params = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_addr = ste_dma_params->tx_rx_addr; } else { snd_dma_params = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); dma_addr = snd_dma_params->addr; } diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index bcf442faff7c..68af2176b19c 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -373,7 +373,7 @@ static int xtfpga_pcm_open(struct snd_soc_component *component, void *p; snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); - p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + p = snd_soc_dai_get_dma_data(asoc_rtd_to_cpu(rtd, 0), substream); runtime->private_data = p; return 0; diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index 60382ec23832..a3a07c0730e6 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -322,7 +322,6 @@ static int zx_spdif_probe(struct platform_device *pdev) zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(zx_spdif->reg_base)) { - dev_err(&pdev->dev, "ioremap failed!\n"); return PTR_ERR(zx_spdif->reg_base); } diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index 0e5a05b25a77..4f787185d630 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -371,7 +371,6 @@ static struct snd_soc_dai_driver zx_tdm_dai = { static int zx_tdm_probe(struct platform_device *pdev) { - struct device *dev = &pdev->dev; struct of_phandle_args out_args; unsigned int dma_reg_offset; struct zx_tdm_info *zx_tdm; @@ -384,7 +383,7 @@ static int zx_tdm_probe(struct platform_device *pdev) if (!zx_tdm) return -ENOMEM; - zx_tdm->dev = dev; + zx_tdm->dev = &pdev->dev; zx_tdm->dai_wclk = devm_clk_get(&pdev->dev, "wclk"); if (IS_ERR(zx_tdm->dai_wclk)) { diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f418..56031026b113 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/card.c b/sound/usb/card.c index 827fb0bc8b56..fd6fd1726ea0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -72,6 +72,7 @@ static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +static char *delayed_register[SNDRV_CARDS]; bool snd_usb_use_vmalloc = true; bool snd_usb_skip_validation; @@ -95,6 +96,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_array(delayed_register, charp, NULL, 0444); +MODULE_PARM_DESC(delayed_register, "Quirk for delayed registration, given by id:iface, e.g. 0123abcd:4."); module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444); @@ -525,6 +528,21 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } +static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +{ + int i; + unsigned int id, inum; + + for (i = 0; i < ARRAY_SIZE(delayed_register); i++) { + if (delayed_register[i] && + sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && + id == chip->usb_id) + return inum != iface; + } + + return false; +} + static const struct usb_device_id usb_audio_ids[]; /* defined below */ /* look for the corresponding quirk */ @@ -662,10 +680,22 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } - /* we are allowed to call snd_card_register() many times */ - err = snd_card_register(chip->card); - if (err < 0) - goto __error; + if (chip->need_delayed_register) { + dev_info(&dev->dev, + "Found post-registration device assignment: %08x:%02x\n", + chip->usb_id, ifnum); + chip->need_delayed_register = false; /* clear again */ + } + + /* we are allowed to call snd_card_register() many times, but first + * check to see if a device needs to skip it or do anything special + */ + if (!snd_usb_registration_quirk(chip, ifnum) && + !check_delayed_register_option(chip, ifnum)) { + err = snd_card_register(chip->card); + if (err < 0) + goto __error; + } if (quirk && quirk->shares_media_device) { /* don't want to fail when snd_media_device_create() fails */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index a48313dfa967..b118cf97607f 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,16 +151,15 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } -/* - * Assume the clock is valid if clock source supports only one single sample - * rate, the terminal is connected directly to it (there is no clock selector) - * and clock type is internal. This is to deal with some Denon DJ controllers - * that always reports that clock is invalid. - */ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, int source_id) { + bool ret = false; + int count; + unsigned char data; + struct usb_device *dev = chip->dev; + if (fmt->protocol == UAC_VERSION_2) { struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); @@ -168,13 +167,51 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, if (!cs_desc) return false; - return (fmt->nr_rates == 1 && - (fmt->clock & 0xff) == cs_desc->bClockID && - (cs_desc->bmAttributes & 0x3) != - UAC_CLOCK_SOURCE_TYPE_EXT); + /* + * Assume the clock is valid if clock source supports only one + * single sample rate, the terminal is connected directly to it + * (there is no clock selector) and clock type is internal. + * This is to deal with some Denon DJ controllers that always + * reports that clock is invalid. + */ + if (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT) + return true; + } + + /* + * MOTU MicroBook IIc + * Sample rate changes takes more than 2 seconds for this device. Clock + * validity request returns false during that period. + */ + if (chip->usb_id == USB_ID(0x07fd, 0x0004)) { + count = 0; + + while ((!ret) && (count < 50)) { + int err; + + msleep(100); + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_CLOCK_VALID << 8, + snd_usb_ctrl_intf(chip) | (source_id << 8), + &data, sizeof(data)); + if (err < 0) { + dev_warn(&dev->dev, + "%s(): cannot get clock validity for id %d\n", + __func__, source_id); + return false; + } + + ret = !!data; + count++; + } } - return false; + return ret; } static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, diff --git a/sound/usb/format.c b/sound/usb/format.c index 9f5cb4ed3a0c..50e1874c847c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -247,6 +247,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } + +/* + * Presonus Studio 1810c supports a limited set of sampling + * rates per altsetting but reports the full set each time. + * If we don't filter out the unsupported rates and attempt + * to configure the card, it will hang refusing to do any + * further audio I/O until a hard reset is performed. + * + * The list of supported rates per altsetting (set of available + * I/O channels) is described in the owner's manual, section 2.2. + */ +static bool s1810c_valid_sample_rate(struct audioformat *fp, + unsigned int rate) +{ + switch (fp->altsetting) { + case 1: + /* All ADAT ports available */ + return rate <= 48000; + case 2: + /* Half of ADAT ports available */ + return (rate == 88200 || rate == 96000); + case 3: + /* Analog I/O only (no S/PDIF nor ADAT) */ + return rate >= 176400; + default: + return false; + } + return false; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -283,6 +313,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* Filter out invalid rates on Presonus Studio 1810c */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + !s1810c_valid_sample_rate(fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) @@ -297,6 +333,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, break; } +skip_rate: /* avoid endless loop */ if (res == 0) break; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 392e5fda680c..047b90595d65 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor { __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[0]; + __u8 baAssocJackID[]; } __attribute__ ((packed)); struct snd_usb_midi_in_endpoint; @@ -1826,6 +1826,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1863,11 +1885,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81b2db0edd5f..721d12130d0c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -292,6 +292,11 @@ static int uac2_ctl_value_size(int val_type) * retrieve a mixer value */ +static inline int mixer_ctrl_intf(struct usb_mixer_interface *mixer) +{ + return get_iface_desc(mixer->hostif)->bInterfaceNumber; +} + static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { @@ -306,7 +311,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, val_len); @@ -354,7 +359,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, size); @@ -479,7 +484,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, return -EIO; while (timeout-- > 0) { - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, @@ -901,6 +906,12 @@ static int parse_term_effect_unit(struct mixer_build *state, struct usb_audio_term *term, void *p1, int id) { + struct uac2_effect_unit_descriptor *d = p1; + int err; + + err = __check_input_term(state, d->bSourceID, term); + if (err < 0) + return err; term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ term->id = id; return 0; @@ -1203,7 +1214,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { usb_audio_err(cval->head.mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip), + cval->head.id, mixer_ctrl_intf(cval->head.mixer), cval->control, cval->head.id); return -EINVAL; } @@ -1422,7 +1433,7 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, if (ret) goto error; - idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + idx = mixer_ctrl_intf(cval->head.mixer) | (cval->head.id << 8); if (cval->head.mixer->protocol == UAC_VERSION_2) { struct uac2_connectors_ctl_blk uac2_conn; @@ -1674,6 +1685,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { @@ -3203,7 +3224,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", - chip->usb_id, snd_usb_ctrl_intf(chip), + chip->usb_id, mixer_ctrl_intf(mixer), mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c237e24f08d9..02b036b2aefb 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 000000000000..6483e47bafd0 --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,595 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com> + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ + +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" +#include "mixer_s1810c.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + u32 a; + u32 b; + u32 fixed1; + u32 fixed2; + u32 c; + u32 d; + u32 e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + u32 fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; + struct mutex data_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a, + u32 b, u32 c, u32 d, u32 e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + u32 *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { { 0 } }; + struct s1810c_state_packet pkt_in = { { 0 } }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + u32 a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes, so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, u32 *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 field = 0; + u32 ctl_idx = (u32) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_id = (pval >> 8) & 0xFF; + u32 ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 state = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + ctl_elem->value.enumerated.item[0] = (int)state; + break; + default: + ctl_elem->value.integer.value[0] = (long)state; + } + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? ret : 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 curval = 0; + u32 newval = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + newval = (u32) ctl_elem->value.enumerated.item[0]; + break; + default: + newval = (u32) ctl_elem->value.integer.value[0]; + } + + if (curval == newval) + goto unlock; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? 0 : 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Preamp On (Mic/Inst)", + "Preamp Off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 Source Type", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom Power On Mic Inputs Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 Source Route", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + kfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL); + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 000000000000..a79a3743cff3 --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm@gmail.com> + */ + +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bd258f1ec2dd..a4e4064f9aee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -357,7 +357,12 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; - case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */ + /* MicroBook IIc */ + if (altsd->bInterfaceClass == USB_CLASS_AUDIO) + return 0; + + /* MicroBook II */ ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/proc.c b/sound/usb/proc.c index ffbf4bd9208c..4174ad11fca6 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -70,7 +70,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); snd_iprintf(buffer, " Format:"); - for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + pcm_for_each_format(fmt) if (fp->formats & pcm_format_to_bits(fmt)) snd_iprintf(buffer, " %s", snd_pcm_format_name(fmt)); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..1c8719292eee 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3472,7 +3472,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, /* MOTU Microbook II */ { - USB_DEVICE(0x07fd, 0x0004), + USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MOTU", .product_name = "MicroBookII", diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7f558f4b4520..86f192a3043d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1252,6 +1252,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Altno settings: + * + * Playback (Interface 1): + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture (Interface 2): + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if ((chip->setup == 0 || chip->setup > 2) && altno != 2) + return 1; + else if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1265,6 +1297,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } @@ -1316,7 +1352,15 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ - return snd_usb_motu_microbookii_boot_quirk(dev); + /* + * For some reason interface 3 with vendor-spec class is + * detected on MicroBook IIc. + */ + if (get_iface_desc(intf->altsetting)->bInterfaceClass == + USB_CLASS_VENDOR_SPEC && + get_iface_desc(intf->altsetting)->bInterfaceNumber < 3) + return snd_usb_motu_microbookii_boot_quirk(dev); + break; } return 0; @@ -1754,5 +1798,47 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, else fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */ + /* + * MaxPacketsOnly attribute is erroneously set in endpoint + * descriptors. As a result this card produces noise with + * all sample rates other than 96 KHz. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; + break; } } + +/* + * registration quirk: + * the registration is skipped if a device matches with the given ID, + * unless the interface reaches to the defined one. This is for delaying + * the registration until the last known interface, so that the card and + * devices appear at the same time. + */ + +struct registration_quirk { + unsigned int usb_id; /* composed via USB_ID() */ + unsigned int interface; /* the interface to trigger register */ +}; + +#define REG_QUIRK_ENTRY(vendor, product, iface) \ + { .usb_id = USB_ID(vendor, product), .interface = (iface) } + +static const struct registration_quirk registration_quirks[] = { + REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ + { 0 } /* terminator */ +}; + +/* return true if skipping registration */ +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) +{ + const struct registration_quirk *q; + + for (q = registration_quirks; q->usb_id; q++) + if (chip->usb_id == q->usb_id) + return iface != q->interface; + + /* Register as normal */ + return false; +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index df0355843a4c..c76cf24a640a 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -51,4 +51,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); +bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); + #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index afd5aa574611..15296f2c902c 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -502,6 +502,9 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, subs = &as->substream[stream]; if (subs->ep_num) continue; + if (snd_device_get_state(chip->card, as->pcm) != + SNDRV_DEV_BUILD) + chip->need_delayed_register = true; err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6fe3ab582ec6..1c892c7f14d7 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -34,6 +34,7 @@ struct snd_usb_audio { unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */ + unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 772f6f3ccbb1..37d290fe9d43 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops = */ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream) { - kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); - usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; + int stream; - kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]); - usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL; + for_each_pcm_streams(stream) { + kfree(usX2Y_substream[stream]); + usX2Y_substream[stream] = NULL; + } } static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm) |