diff options
Diffstat (limited to 'sound')
85 files changed, 1082 insertions, 1553 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b03a638b420c..279e24f61305 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1552,6 +1552,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); + } else { + runtime->status->state = SNDRV_PCM_STATE_SETUP; } break; case SNDRV_PCM_STATE_RUNNING: diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 0d580186ef1a..5cc356db5351 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -33,7 +33,7 @@ */ #define MAX_MIDI_RX_BLOCKS 8 -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 µs */ +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 @@ -78,7 +78,7 @@ static void pcm_period_tasklet(unsigned long data); int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, enum cip_flags flags) { - s->unit = fw_unit_get(unit); + s->unit = unit; s->direction = dir; s->flags = flags; s->context = ERR_PTR(-1); @@ -102,7 +102,6 @@ void amdtp_stream_destroy(struct amdtp_stream *s) { WARN_ON(amdtp_stream_running(s)); mutex_destroy(&s->mutex); - fw_unit_put(s->unit); } EXPORT_SYMBOL(amdtp_stream_destroy); diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index fc19c99654aa..611b7dae7ee5 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -116,11 +116,22 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; + snd_bebob_stream_destroy_duplex(bebob); + fw_unit_put(bebob->unit); + + kfree(bebob->maudio_special_quirk); + if (bebob->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(bebob->card_index, devices_used); @@ -205,7 +216,7 @@ bebob_probe(struct fw_unit *unit, card->private_free = bebob_card_free; bebob->card = card; - bebob->unit = unit; + bebob->unit = fw_unit_get(unit); bebob->spec = spec; mutex_init(&bebob->mutex); spin_lock_init(&bebob->lock); @@ -306,10 +317,11 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; - kfree(bebob->maudio_special_quirk); + /* Awake bus-reset waiters. */ + if (!completion_done(&bebob->bus_reset)) + complete_all(&bebob->bus_reset); - snd_bebob_stream_destroy_duplex(bebob); - snd_card_disconnect(bebob->card); + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(bebob->card); } diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 0ebcabfdc7ce..98e4fc8121a1 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -410,8 +410,6 @@ break_both_connections(struct snd_bebob *bebob) static void destroy_both_connections(struct snd_bebob *bebob) { - break_both_connections(bebob); - cmp_connection_destroy(&bebob->in_conn); cmp_connection_destroy(&bebob->out_conn); } @@ -712,22 +710,16 @@ void snd_bebob_stream_update_duplex(struct snd_bebob *bebob) mutex_unlock(&bebob->mutex); } +/* + * This function should be called before starting streams or after stopping + * streams. + */ void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob) { - mutex_lock(&bebob->mutex); - - amdtp_stream_pcm_abort(&bebob->rx_stream); - amdtp_stream_pcm_abort(&bebob->tx_stream); - - amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_stop(&bebob->tx_stream); - amdtp_stream_destroy(&bebob->rx_stream); amdtp_stream_destroy(&bebob->tx_stream); destroy_both_connections(bebob); - - mutex_unlock(&bebob->mutex); } /* diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index fa9cf761b610..07dbd01d7a6b 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -311,14 +311,21 @@ end: return err; } +/* + * This function should be called before starting streams or after stopping + * streams. + */ static void destroy_stream(struct snd_dice *dice, struct amdtp_stream *stream) { - amdtp_stream_destroy(stream); + struct fw_iso_resources *resources; if (stream == &dice->tx_stream) - fw_iso_resources_destroy(&dice->tx_resources); + resources = &dice->tx_resources; else - fw_iso_resources_destroy(&dice->rx_resources); + resources = &dice->rx_resources; + + amdtp_stream_destroy(stream); + fw_iso_resources_destroy(resources); } int snd_dice_stream_init_duplex(struct snd_dice *dice) @@ -332,6 +339,8 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice) goto end; err = init_stream(dice, &dice->rx_stream); + if (err < 0) + destroy_stream(dice, &dice->tx_stream); end: return err; } @@ -340,10 +349,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { snd_dice_transaction_clear_enable(dice); - stop_stream(dice, &dice->tx_stream); destroy_stream(dice, &dice->tx_stream); - - stop_stream(dice, &dice->rx_stream); destroy_stream(dice, &dice->rx_stream); dice->substreams_counter = 0; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 90d8f40ff727..70a111d7f428 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -226,11 +226,20 @@ static void dice_card_strings(struct snd_dice *dice) strcpy(card->mixername, "DICE"); } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void dice_card_free(struct snd_card *card) { struct snd_dice *dice = card->private_data; + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); + fw_unit_put(dice->unit); + mutex_destroy(&dice->mutex); } @@ -251,7 +260,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice = card->private_data; dice->card = card; - dice->unit = unit; + dice->unit = fw_unit_get(unit); card->private_free = dice_card_free; spin_lock_init(&dice->lock); @@ -305,10 +314,7 @@ static void dice_remove(struct fw_unit *unit) { struct snd_dice *dice = dev_get_drvdata(&unit->device); - snd_card_disconnect(dice->card); - - snd_dice_stream_destroy_duplex(dice); - + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(dice->card); } diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 3e2ed8e82cbc..2682e7e3e5c9 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -173,11 +173,23 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); + fw_unit_put(efw->unit); + + kfree(efw->resp_buf); + if (efw->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(efw->card_index, devices_used); @@ -185,7 +197,6 @@ efw_card_free(struct snd_card *card) } mutex_destroy(&efw->mutex); - kfree(efw->resp_buf); } static int @@ -218,7 +229,7 @@ efw_probe(struct fw_unit *unit, card->private_free = efw_card_free; efw->card = card; - efw->unit = unit; + efw->unit = fw_unit_get(unit); mutex_init(&efw->mutex); spin_lock_init(&efw->lock); init_waitqueue_head(&efw->hwdep_wait); @@ -289,10 +300,7 @@ static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - - snd_card_disconnect(efw->card); + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(efw->card); } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 4f440e163667..c55db1bddc80 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -100,17 +100,22 @@ end: return err; } +/* + * This function should be called before starting the stream or after stopping + * the streams. + */ static void destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) { - stop_stream(efw, stream); - - amdtp_stream_destroy(stream); + struct cmp_connection *conn; if (stream == &efw->tx_stream) - cmp_connection_destroy(&efw->out_conn); + conn = &efw->out_conn; else - cmp_connection_destroy(&efw->in_conn); + conn = &efw->in_conn; + + amdtp_stream_destroy(stream); + cmp_connection_destroy(&efw->out_conn); } static int @@ -319,12 +324,8 @@ void snd_efw_stream_update_duplex(struct snd_efw *efw) void snd_efw_stream_destroy_duplex(struct snd_efw *efw) { - mutex_lock(&efw->mutex); - destroy_stream(efw, &efw->rx_stream); destroy_stream(efw, &efw->tx_stream); - - mutex_unlock(&efw->mutex); } void snd_efw_stream_lock_changed(struct snd_efw *efw) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index bda845afb470..29ccb3637164 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -337,6 +337,10 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw, stop_stream(oxfw, stream); } +/* + * This function should be called before starting the stream or after stopping + * the streams. + */ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream) { @@ -347,8 +351,6 @@ void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, else conn = &oxfw->in_conn; - stop_stream(oxfw, stream); - amdtp_stream_destroy(stream); cmp_connection_destroy(conn); } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 60e5cad0531a..8c6ce019f437 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -104,11 +104,23 @@ end: return err; } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ static void oxfw_card_free(struct snd_card *card) { struct snd_oxfw *oxfw = card->private_data; unsigned int i; + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + + fw_unit_put(oxfw->unit); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { kfree(oxfw->tx_stream_formats[i]); kfree(oxfw->rx_stream_formats[i]); @@ -136,7 +148,7 @@ static int oxfw_probe(struct fw_unit *unit, oxfw = card->private_data; oxfw->card = card; mutex_init(&oxfw->mutex); - oxfw->unit = unit; + oxfw->unit = fw_unit_get(unit); oxfw->device_info = (const struct device_info *)id->driver_data; spin_lock_init(&oxfw->lock); init_waitqueue_head(&oxfw->hwdep_wait); @@ -212,12 +224,7 @@ static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - snd_card_disconnect(oxfw->card); - - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - + /* No need to wait for releasing card object in this context. */ snd_card_free_when_closed(oxfw->card); } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9f..a2ce773bdc62 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -961,7 +961,6 @@ static int azx_alloc_cmd_io(struct azx *chip) dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); return err; } -EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); static void azx_init_cmd_io(struct azx *chip) { @@ -1026,7 +1025,6 @@ static void azx_init_cmd_io(struct azx *chip) azx_writeb(chip, RIRBCTL, AZX_RBCTL_DMA_EN | AZX_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_init_cmd_io); static void azx_free_cmd_io(struct azx *chip) { @@ -1036,7 +1034,6 @@ static void azx_free_cmd_io(struct azx *chip) azx_writeb(chip, CORBCTL, 0); spin_unlock_irq(&chip->reg_lock); } -EXPORT_SYMBOL_GPL(azx_free_cmd_io); static unsigned int azx_command_addr(u32 cmd) { @@ -1316,7 +1313,6 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val) else return azx_corb_send_cmd(bus, val); } -EXPORT_SYMBOL_GPL(azx_send_cmd); /* get a response */ static unsigned int azx_get_response(struct hda_bus *bus, @@ -1330,7 +1326,6 @@ static unsigned int azx_get_response(struct hda_bus *bus, else return azx_rirb_get_response(bus, addr); } -EXPORT_SYMBOL_GPL(azx_get_response); #ifdef CONFIG_SND_HDA_DSP_LOADER /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36d2f20db7a4..4ca3d5d02436 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1966,7 +1966,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6d36c5b78805..87eff3173ce9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -79,6 +79,7 @@ enum { STAC_ALIENWARE_M17X, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, + STAC_92HD73XX_ASUS_MOBO, STAC_92HD73XX_MODELS }; @@ -1911,7 +1912,18 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK] = { .type = HDA_FIXUP_PINS, .v.pins = stac92hd89xx_hp_z1_g2_right_mic_jack_pin_configs, - } + }, + [STAC_92HD73XX_ASUS_MOBO] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* enable 5.1 and SPDIF out */ + { 0x0c, 0x01014411 }, + { 0x0d, 0x01014410 }, + { 0x0e, 0x01014412 }, + { 0x22, 0x014b1180 }, + { } + } + }, }; static const struct hda_model_fixup stac92hd73xx_models[] = { @@ -1923,6 +1935,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -1975,6 +1988,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), + SND_PCI_QUIRK(PCI_VENDOR_ID_ASUSTEK, 0x83f8, "ASUS AT4NM10", + STAC_92HD73XX_ASUS_MOBO), {} /* terminator */ }; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index fb0b7e8b08ff..841d05946b88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f5ad214663f9..8de836165cf2 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,8 +46,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> - #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) int ret; if (!np) { - if (!(machine_is_at91sam9g20ek() || - machine_is_at91sam9g20ek_2mmc())) - return -ENODEV; + return -ENODEV; } ret = atmel_ssc_set_audio(0); @@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* Parse device node info */ - if (np) { - ret = snd_soc_of_parse_card_name(card, "atmel,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "atmel,audio-routing"); - if (ret) - goto err; - - /* Parse codec info */ - at91sam9g20ek_dai.codec_name = NULL; - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { - dev_err(&pdev->dev, "codec info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.codec_of_node = codec_np; - - /* Parse dai and platform info */ - at91sam9g20ek_dai.cpu_dai_name = NULL; - at91sam9g20ek_dai.platform_name = NULL; - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { - dev_err(&pdev->dev, "dai and pcm info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.cpu_of_node = cpu_np; - at91sam9g20ek_dai.platform_of_node = cpu_np; - - of_node_put(codec_np); - of_node_put(cpu_np); + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "atmel,audio-routing"); + if (ret) + goto err; + + /* Parse codec info */ + at91sam9g20ek_dai.codec_name = NULL; + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "codec info missing\n"); + return -EINVAL; + } + at91sam9g20ek_dai.codec_of_node = codec_np; + + /* Parse dai and platform info */ + at91sam9g20ek_dai.cpu_dai_name = NULL; + at91sam9g20ek_dai.platform_name = NULL; + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "dai and pcm info missing\n"); + return -EINVAL; } + at91sam9g20ek_dai.cpu_of_node = cpu_np; + at91sam9g20ek_dai.platform_of_node = cpu_np; + + of_node_put(codec_np); + of_node_put(cpu_np); ret = snd_soc_register_card(card); if (ret) { diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 7b7fbcd49e5e..c7cd60f009e9 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97 config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C select SND_EP93XX_SOC_I2S select SND_SOC_TLV320AIC23_I2C help diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e109..ea9f0e31f9d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 1806333ea29e..bf3e933ee895 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -12,11 +12,19 @@ * max98357a.c -- MAX98357A ALSA SoC Codec driver */ -#include <linux/module.h> +#include <linux/device.h> +#include <linux/err.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> #include <sound/soc.h> - -#define DRV_NAME "max98357a" +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) @@ -77,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "HiFi", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -117,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -125,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -135,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f201a08f..4b5f1fe9be97 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -54,6 +54,9 @@ struct pcm512x_priv { int pll_d; int pll_p; unsigned long real_pll; + unsigned long overclock_pll; + unsigned long overclock_dac; + unsigned long overclock_dsp; }; /* @@ -224,6 +227,90 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } } +static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_pll; + return 0; +} + +static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_pll = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dsp; + return 0; +} + +static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dsp = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dac; + return 0; +} + +static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dac = ucontrol->value.integer.value[0]; + return 0; +} + static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); @@ -328,6 +415,13 @@ SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), + +SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0, + pcm512x_overclock_pll_get, pcm512x_overclock_pll_put), +SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put), +SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dac_get, pcm512x_overclock_dac_put), }; static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { @@ -346,6 +440,45 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x) +{ + return 25000000 + 25000000 * pcm512x->overclock_pll / 100; +} + +static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x) +{ + return 50000000 + 50000000 * pcm512x->overclock_dsp / 100; +} + +static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x, + unsigned long rate) +{ + return rate + rate * pcm512x->overclock_dac / 100; +} + +static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x) +{ + if (!pcm512x->pll_out) + return 25000000; + return pcm512x_pll_max(pcm512x); +} + +static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, + unsigned long dac_rate) +{ + /* + * If the DAC is not actually overclocked, use the good old + * NCP target rate... + */ + if (dac_rate <= 6144000) + return 1536000; + /* + * ...but if the DAC is in fact overclocked, bump the NCP target + * rate to get the recommended dividers even when overclocking. + */ + return pcm512x_dac_max(pcm512x, 1536000); +} + static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, @@ -359,6 +492,7 @@ static const struct snd_pcm_hw_constraint_list constraints_slave = { static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct pcm512x_priv *pcm512x = rule->private; struct snd_interval ranges[2]; int frame_size; @@ -377,7 +511,7 @@ static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, */ memset(ranges, 0, sizeof(ranges)); ranges[0].min = 8000; - ranges[0].max = 25000000 / frame_size / 2; + ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2; ranges[1].min = DIV_ROUND_UP(16000000, frame_size); ranges[1].max = 384000; break; @@ -408,7 +542,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, return snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, - NULL, + pcm512x, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -517,6 +651,8 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, unsigned long bclk_rate) { struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); unsigned long sck_rate; int pow2; @@ -527,9 +663,10 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, * as many factors of 2 as possible, as that makes it easier * to find a fast DAC rate */ - pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate); for (; pow2; pow2 >>= 1) { - sck_rate = rounddown(25000000, bclk_rate * pow2); + sck_rate = rounddown(pcm512x_pll_max(pcm512x), + bclk_rate * pow2); if (sck_rate >= 16000000) break; } @@ -678,7 +815,7 @@ static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, return 0; /* futile, quit early */ /* run DAC no faster than 6144000 Hz */ - for (dac_rate = rounddown(6144000, osr_rate); + for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate); dac_rate; dac_rate -= osr_rate) { @@ -805,7 +942,7 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, osr_rate = 16 * sample_rate; /* run DSP no faster than 50 MHz */ - dsp_div = mck_rate > 50000000 ? 2 : 1; + dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1; dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); if (dac_rate) { @@ -836,7 +973,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = pllin_rate; } else { /* run DAC no faster than 6144000 Hz */ - unsigned long dac_mul = 6144000 / osr_rate; + unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000) + / osr_rate; unsigned long sck_mul = sck_rate / osr_rate; for (; dac_mul; dac_mul--) { @@ -863,28 +1001,30 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = sck_rate; } + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); if (dac_div > 128) { dev_err(dev, "Failed to find DAC divider\n"); return -EINVAL; } + dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); - if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + ncp_div = DIV_ROUND_CLOSEST(dac_rate, + pcm512x_ncp_target(pcm512x, dac_rate)); + if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ - ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + ncp_div = DIV_ROUND_UP(dac_rate, 2048000); if (ncp_div > 128) { dev_err(dev, "Failed to find NCP divider\n"); return -EINVAL; } } - osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); - if (osr_div > 128) { - dev_err(dev, "Failed to find OSR divider\n"); - return -EINVAL; - } - idac = mck_rate / (dsp_div * sample_rate); ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); @@ -937,11 +1077,11 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, return ret; } - if (sample_rate <= 48000) + if (sample_rate <= pcm512x_dac_max(pcm512x, 48000)) fssp = PCM512x_FSSP_48KHZ; - else if (sample_rate <= 96000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000)) fssp = PCM512x_FSSP_96KHZ; - else if (sample_rate <= 192000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000)) fssp = PCM512x_FSSP_192KHZ; else fssp = PCM512x_FSSP_384KHZ; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index e1a4a45c57e2..fd102613d20d 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg) case RT5670_ADC_EQ_CTRL1: case RT5670_EQ_CTRL1: case RT5670_ALC_CTRL_1: - case RT5670_IRQ_CTRL1: case RT5670_IRQ_CTRL2: case RT5670_INT_IRQ_ST: case RT5670_IL_CMD: @@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..fb9c20eace3f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, - { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, @@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, - { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, @@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, - { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, - { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" }, { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, { "DAC2 L Mux", "OB 2", "OutBound2" }, - { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, - { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" }, { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, @@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, - { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, - { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" }, { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, { "DAC3 L Mux", "OB 4", "OutBound4" }, - { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, - { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" }, { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, { "DAC3 R Mux", "OB 5", "OutBound5" }, - { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, { "DAC4 L Mux", "OB 6", "OutBound6" }, - { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e41809..1e5d2643c286 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -783,19 +783,21 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); } -static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +static int sn95031_get_headset_state(struct snd_soc_codec *codec, + struct snd_soc_jack *mfld_jack) { - int micbias = sn95031_get_mic_bias(mfld_jack->codec); + int micbias = sn95031_get_mic_bias(codec); int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(mfld_jack->codec); + sn95031_enable_jack_btn(codec); return jack_type; } -void sn95031_jack_detection(struct mfld_jack_data *jack_data) +void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data) { unsigned int status; unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; @@ -809,11 +811,11 @@ void sn95031_jack_detection(struct mfld_jack_data *jack_data) status = SND_JACK_HEADSET | SND_JACK_BTN_1; } else if (jack_data->intr_id & 0x4) { pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(jack_data->mfld_jack); + status = sn95031_get_headset_state(codec, jack_data->mfld_jack); } else if (jack_data->intr_id & 0x8) { pr_debug("headset or headphones removed\n"); status = 0; - sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + sn95031_disable_jack_btn(codec); } else { pr_err("unidentified interrupt\n"); return; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 20376d234fb8..7651fe4e6a45 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -127,6 +127,7 @@ struct mfld_jack_data { struct snd_soc_jack *mfld_jack; }; -extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); +extern void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data); #endif diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 3a1343fa109b..007a0e3bc273 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = { }; static const struct regmap_range sta32x_write_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_read_regs_range[] = { - regmap_reg_range(STA32X_CONFA, STA32X_AUTO2), - regmap_reg_range(STA32X_C1CFG, STA32X_FDRC2), + regmap_reg_range(STA32X_CONFA, STA32X_FDRC2), }; static const struct regmap_range sta32x_volatile_regs_range[] = { diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769e..669e3228241e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd2708..dfb4ff5cc9ea 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 2b81ca418d2a..3736d9aabc56 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,14 +1,16 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI" + tristate depends on ARCH_DAVINCI + select SND_EDMA_SOC config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" - depends on SOC_AM33XX || SOC_AM43XX + tristate "SoC Audio for Texas Instruments chips using eDMA" + depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. The following line of SoCs are supported by this platform driver: + - daVinci devices - AM335x - AM437x/AM438x @@ -17,7 +19,7 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + depends on SND_OMAP_SOC || SND_EDMA_SOC help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -45,7 +47,7 @@ config SND_AM33XX_SOC_EVM config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_DAVINCI_SOC && I2C + depends on SND_EDMA_SOC && I2C depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_GENERIC_EVM help @@ -73,7 +75,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF @@ -82,7 +84,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help @@ -91,7 +93,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 09bf2ba92d38..f883933c1a19 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,11 +1,9 @@ # DAVINCI Platform Support -snd-soc-davinci-objs := davinci-pcm.o snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o -obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 15fb28fc8e1b..56cb4d95637d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -23,8 +23,9 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" @@ -122,7 +123,8 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { struct davinci_mcbsp_dev { struct device *dev; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 @@ -419,8 +421,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length, master; unsigned int rcr, xcr, srgr, clk_div, freq, framesize; @@ -532,8 +532,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - dma_params->acnt = dma_params->data_type = data_type[fmt]; - dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; switch (master) { @@ -600,15 +598,6 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -620,7 +609,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -630,7 +618,18 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { }; +static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_i2s_dai = { + .probe = davinci_i2s_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -651,11 +650,9 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - enum dma_event_q asp_chan_q = EVENTQ_0; - enum dma_event_q ram_chan_q = EVENTQ_1; + int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -676,22 +673,6 @@ static int davinci_i2s_probe(struct platform_device *pdev) GFP_KERNEL); if (!dev) return -ENOMEM; - if (pdata) { - dev->enable_channel_combine = pdata->enable_channel_combine; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = - pdata->sram_size_playback; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = - pdata->sram_size_capture; - dev->clk_input_pin = pdata->clk_input_pin; - dev->i2s_accurate_sck = pdata->i2s_accurate_sck; - asp_chan_q = pdata->asp_chan_q; - ram_chan_q = pdata->ram_chan_q; - } - - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) @@ -705,10 +686,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -718,7 +699,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -726,9 +709,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; - dev->dev = &pdev->dev; + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, @@ -736,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err_unregister_component; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155a5011..0c882995a357 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/platform_data/davinci_asp.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -36,7 +37,6 @@ #include <sound/dmaengine_pcm.h> #include <sound/omap-pcm.h> -#include "davinci-pcm.h" #include "edma-pcm.h" #include "davinci-mcasp.h" @@ -65,7 +65,6 @@ struct davinci_mcasp_context { }; struct davinci_mcasp { - struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; @@ -82,6 +81,7 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; + int dma_request[2]; int sysclk_freq; bool bclk_master; @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -631,7 +643,6 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int period_words, int channels) { - struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream]; struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream]; int i; u8 tx_ser = 0; @@ -699,10 +710,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * For example if three serializers are enabled the DMA * need to transfer three words per DMA request. */ - dma_params->fifo_level = active_serializers; dma_data->maxburst = active_serializers; } else { - dma_params->fifo_level = 0; dma_data->maxburst = 0; } return 0; @@ -734,7 +743,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, /* Configure the burst size for platform drivers */ if (numevt == 1) numevt = 0; - dma_params->fifo_level = numevt; dma_data->maxburst = numevt; return 0; @@ -860,8 +868,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - struct davinci_pcm_dma_params *dma_params = - &mcasp->dma_params[substream->stream]; int word_length; int channels = params_channels(params); int period_size = params_period_size(params); @@ -902,31 +908,26 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - dma_params->data_type = 3; word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: - dma_params->data_type = 4; word_length = 24; break; case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; word_length = 32; break; @@ -935,11 +936,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level) - dma_params->acnt = 4; - else - dma_params->acnt = dma_params->data_type; - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1043,17 +1039,8 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version >= MCASP_VERSION_3) { - /* Using dmaengine PCM */ - dai->playback_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - } else { - /* Using davinci-pcm */ - dai->playback_dma_data = mcasp->dma_params; - dai->capture_dma_data = mcasp->dma_params; - } + dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; return 0; } @@ -1172,28 +1159,24 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_4, }; @@ -1370,12 +1353,12 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_params; struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; + int *dma; int irq; int ret; @@ -1509,59 +1492,45 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->tx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->tx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->tx_dma_channel; + *dma = pdata->tx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "tx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; /* RX is not valid in DIT mode */ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->rx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->rx_dma_channel; + *dma = pdata->rx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "rx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; } if (mcasp->version < MCASP_VERSION_3) { @@ -1584,17 +1553,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { -#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_DAVINCI_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - ret = davinci_soc_platform_register(&pdev->dev); - break; -#endif #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_1: + case MCASP_VERSION_2: case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); break; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c deleted file mode 100644 index 7809e9d935fc..000000000000 --- a/sound/soc/davinci/davinci-pcm.c +++ /dev/null @@ -1,861 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> - * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> - * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/kernel.h> -#include <linux/genalloc.h> -#include <linux/platform_data/edma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "davinci-pcm.h" - -#ifdef DEBUG -static void print_buf_info(int slot, char *name) -{ - struct edmacc_param p; - if (slot < 0) - return; - edma_read_slot(slot, &p); - printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", - name, slot, p.opt, p.src, p.a_b_cnt, p.dst); - printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", - p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); -} -#else -static void print_buf_info(int slot, char *name) -{ -} -#endif - -static struct snd_pcm_hardware pcm_hardware_playback = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware pcm_hardware_capture = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -/* - * How ping/pong works.... - * - * Playback: - * ram_params - copys 2*ping_size from start of SDRAM to iram, - * links to ram_link2 - * ram_link2 - copys rest of SDRAM to iram in ping_size units, - * links to ram_link - * ram_link - copys entire SDRAM to iram in ping_size uints, - * links to self - * - * asp_params - same as asp_link[0] - * asp_link[0] - copys from lower half of iram to asp port - * links to asp_link[1], triggers iram copy event on completion - * asp_link[1] - copys from upper half of iram to asp port - * links to asp_link[0], triggers iram copy event on completion - * triggers interrupt only needed to let upper SOC levels update position - * in stream on completion - * - * When playback is started: - * ram_params started - * asp_params started - * - * Capture: - * ram_params - same as ram_link, - * links to ram_link - * ram_link - same as playback - * links to self - * - * asp_params - same as playback - * asp_link[0] - same as playback - * asp_link[1] - same as playback - * - * When capture is started: - * asp_params started - */ -struct davinci_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ - int asp_channel; /* Master DMA channel */ - int asp_link[2]; /* asp parameter link channel, ping/pong */ - struct davinci_pcm_dma_params *params; /* DMA params */ - int ram_channel; - int ram_link; - int ram_link2; - struct edmacc_param asp_params; - struct edmacc_param ram_params; -}; - -static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; -} -/* - * Not used with ping/pong - */ -static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - unsigned short src_bidx, dst_bidx; - unsigned short src_cidx, dst_cidx; - unsigned int data_type; - unsigned short acnt; - unsigned int count; - unsigned int fifo_level; - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - fifo_level = prtd->params->fifo_level; - - pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, - period_size); - - data_type = prtd->params->data_type; - count = period_size / data_type; - if (fifo_level) - count /= fifo_level; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = prtd->params->dma_addr; - src_bidx = data_type; - dst_bidx = 4; - src_cidx = data_type * fifo_level; - dst_cidx = 0; - } else { - src = prtd->params->dma_addr; - dst = dma_pos; - src_bidx = 0; - dst_bidx = data_type; - src_cidx = 0; - dst_cidx = data_type * fifo_level; - } - - acnt = prtd->params->acnt; - edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); - edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); - edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); - - if (!fifo_level) - edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, - ASYNC); - else - edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level, - count, fifo_level, - ABSYNC); -} - -static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) -{ - struct snd_pcm_substream *substream = data; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - print_buf_info(prtd->ram_channel, "i ram_channel"); - pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - - if (unlikely(ch_status != EDMA_DMA_COMPLETE)) - return; - - if (snd_pcm_running(substream)) { - spin_lock(&prtd->lock); - if (prtd->ram_channel < 0) { - /* No ping/pong must fix up link dma data*/ - davinci_pcm_enqueue_dma(substream); - } - davinci_pcm_period_elapsed(substream); - spin_unlock(&prtd->lock); - snd_pcm_period_elapsed(substream); - } -} - -#ifdef CONFIG_GENERIC_ALLOCATOR -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct snd_dma_buffer *iram_dma = NULL; - dma_addr_t iram_phys = 0; - void *iram_virt = NULL; - - if (buf->private_data || !size) - return 0; - - ppcm->period_bytes_max = size; - iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys); - if (!iram_virt) - goto exit1; - iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); - if (!iram_dma) - goto exit2; - iram_dma->area = iram_virt; - iram_dma->addr = iram_phys; - memset(iram_dma->area, 0, size); - iram_dma->bytes = size; - buf->private_data = iram_dma; - return 0; -exit2: - if (iram_virt) - gen_pool_free(sram_pool, (unsigned)iram_virt, size); -exit1: - return -ENOMEM; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - - gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); -} -#else -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - return 0; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ -} -#endif - -/* - * Only used with ping/pong. - * This is called after runtime->dma_addr, period_bytes and data_type are valid - */ -static int ping_pong_dma_setup(struct snd_pcm_substream *substream) -{ - unsigned short ram_src_cidx, ram_dst_cidx; - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - struct snd_dma_buffer *iram_dma = - (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct davinci_pcm_dma_params *params = prtd->params; - unsigned int data_type = params->data_type; - unsigned int acnt = params->acnt; - /* divide by 2 for ping/pong */ - unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - unsigned int fifo_level = prtd->params->fifo_level; - unsigned int count; - if ((data_type == 0) || (data_type > 4)) { - printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_addr_t asp_src_pong = iram_dma->addr + ping_size; - ram_src_cidx = ping_size; - ram_dst_cidx = -ping_size; - edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_src_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } else { - dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; - ram_src_cidx = -ping_size; - ram_dst_cidx = ping_size; - edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - - edma_set_dest_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_dest_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } - - if (!fifo_level) { - count = ping_size / data_type; - edma_set_transfer_params(prtd->asp_link[0], acnt, count, - 1, 0, ASYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, count, - 1, 0, ASYNC); - } else { - count = ping_size / (data_type * fifo_level); - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, ABSYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, - count, fifo_level, ABSYNC); - } - - edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); - edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); - edma_set_transfer_params(prtd->ram_link, ping_size, 2, - runtime->periods, 2, ASYNC); - - /* init master params */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_read_slot(prtd->ram_link, &prtd->ram_params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - struct edmacc_param p_ram; - /* Copy entire iram buffer before playback started */ - prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); - /* 0 dst_bidx */ - prtd->ram_params.src_dst_bidx = (ping_size << 1); - /* 0 dst_cidx */ - prtd->ram_params.src_dst_cidx = (ping_size << 1); - prtd->ram_params.ccnt = 1; - - /* Skip 1st period */ - edma_read_slot(prtd->ram_link, &p_ram); - p_ram.src += (ping_size << 1); - p_ram.ccnt -= 1; - edma_write_slot(prtd->ram_link2, &p_ram); - /* - * When 1st started, ram -> iram dma channel will fill the - * entire iram. Then, whenever a ping/pong asp buffer finishes, - * 1/2 iram will be filled. - */ - prtd->ram_params.link_bcntrld = - EDMA_CHAN_SLOT(prtd->ram_link2) << 5; - } - return 0; -} - -/* 1 asp tx or rx channel using 2 parameter channels - * 1 ram to/from iram channel using 1 parameter channel - * - * Playback - * ram copy channel kicks off first, - * 1st ram copy of entire iram buffer completion kicks off asp channel - * asp tcc always kicks off ram copy of 1/2 iram buffer - * - * Record - * asp channel starts, tcc kicks off ram copy - */ -static int request_ping_pong(struct snd_pcm_substream *substream, - struct davinci_runtime_data *prtd, - struct snd_dma_buffer *iram_dma) -{ - dma_addr_t asp_src_ping; - dma_addr_t asp_dst_ping; - int ret; - struct davinci_pcm_dma_params *params = prtd->params; - - /* Request ram master channel */ - ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, - davinci_pcm_dma_irq, substream, - prtd->params->ram_chan_q); - if (ret < 0) - goto exit1; - - /* Request ram link channel */ - ret = prtd->ram_link = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - ret = prtd->asp_link[1] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit3; - - prtd->ram_link2 = -1; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = prtd->ram_link2 = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit4; - } - /* circle ping-pong buffers */ - edma_link(prtd->asp_link[0], prtd->asp_link[1]); - edma_link(prtd->asp_link[1], prtd->asp_link[0]); - /* circle ram buffers */ - edma_link(prtd->ram_link, prtd->ram_link); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - asp_src_ping = iram_dma->addr; - asp_dst_ping = params->dma_addr; /* fifo */ - } else { - asp_src_ping = params->dma_addr; /* fifo */ - asp_dst_ping = iram_dma->addr; - } - /* ping */ - edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[0], 0, 0); - edma_set_dest_index(prtd->asp_link[0], 0, 0); - - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - - /* pong */ - edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[1], 0, 0); - edma_set_dest_index(prtd->asp_link[1], 0, 0); - - edma_read_slot(prtd->asp_link[1], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); - /* interrupt after every pong completion */ - prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[1], &prtd->asp_params); - - /* ram */ - edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," - "for asp:%u %u %u\n", __func__, - prtd->ram_channel, prtd->ram_link, prtd->ram_link2, - prtd->asp_channel, prtd->asp_link[0], - prtd->asp_link[1]); - return 0; -exit4: - edma_free_channel(prtd->asp_link[1]); - prtd->asp_link[1] = -1; -exit3: - edma_free_channel(prtd->ram_link); - prtd->ram_link = -1; -exit2: - edma_free_channel(prtd->ram_channel); - prtd->ram_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) -{ - struct snd_dma_buffer *iram_dma; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct davinci_pcm_dma_params *params = prtd->params; - int ret; - - if (!params) - return -ENODEV; - - /* Request asp master DMA channel */ - ret = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, - prtd->params->asp_chan_q); - if (ret < 0) - goto exit1; - - /* Request asp link channels */ - ret = prtd->asp_link[0] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - if (iram_dma) { - if (request_ping_pong(substream, prtd, iram_dma) == 0) - return 0; - printk(KERN_WARNING "%s: dma channel allocation failed," - "not using sram\n", __func__); - } - - /* Issue transfer completion IRQ when the channel completes a - * transfer, then always reload from the same slot (by a kind - * of loopback link). The completion IRQ handler will update - * the reload slot with a new buffer. - * - * REVISIT save p_ram here after setting up everything except - * the buffer and its length (ccnt) ... use it as a template - * so davinci_pcm_enqueue_dma() takes less time in IRQ. - */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt |= TCINTEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - return 0; -exit2: - edma_free_channel(prtd->asp_channel); - prtd->asp_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - edma_start(prtd->asp_channel); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - prtd->ram_channel >= 0) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_resume(prtd->asp_channel); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_pause(prtd->asp_channel); - break; - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static int davinci_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - davinci_pcm_period_reset(substream); - if (prtd->ram_channel >= 0) { - int ret = ping_pong_dma_setup(substream); - if (ret < 0) - return ret; - - edma_write_slot(prtd->ram_channel, &prtd->ram_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - - print_buf_info(prtd->ram_channel, "ram_channel"); - print_buf_info(prtd->ram_link, "ram_link"); - print_buf_info(prtd->ram_link2, "ram_link2"); - print_buf_info(prtd->asp_channel, "asp_channel"); - print_buf_info(prtd->asp_link[0], "asp_link[0]"); - print_buf_info(prtd->asp_link[1], "asp_link[1]"); - - /* - * There is a phase offset of 2 periods between the position - * used by dma setup and the position reported in the pointer - * function. - * - * The phase offset, when not using ping-pong buffers, is due to - * the two consecutive calls to davinci_pcm_enqueue_dma() below. - * - * Whereas here, with ping-pong buffers, the phase is due to - * there being an entire buffer transfer complete before the - * first dma completion event triggers davinci_pcm_dma_irq(). - */ - davinci_pcm_period_elapsed(substream); - davinci_pcm_period_elapsed(substream); - - return 0; - } - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - return 0; -} - -static snd_pcm_uframes_t -davinci_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - unsigned int offset; - int asp_count; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - - /* - * There is a phase offset of 2 periods between the position used by dma - * setup and the position reported in the pointer function. Either +2 in - * the dma setup or -2 here in the pointer function (with wrapping, - * both) accounts for this offset -- choose the latter since it makes - * the first-time setup clearer. - */ - spin_lock(&prtd->lock); - asp_count = prtd->period - 2; - spin_unlock(&prtd->lock); - - if (asp_count < 0) - asp_count += runtime->periods; - asp_count *= period_size; - - offset = bytes_to_frames(runtime, asp_count); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int davinci_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd; - struct snd_pcm_hardware *ppcm; - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa; - struct davinci_pcm_dma_params *params; - - pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!pa) - return -ENODEV; - params = &pa[substream->stream]; - - ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); - snd_soc_set_runtime_hwparams(substream, ppcm); - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - prtd->params = params; - prtd->asp_channel = -1; - prtd->asp_link[0] = prtd->asp_link[1] = -1; - prtd->ram_channel = -1; - prtd->ram_link = -1; - prtd->ram_link2 = -1; - - runtime->private_data = prtd; - - ret = davinci_pcm_dma_request(substream); - if (ret) { - printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); - kfree(prtd); - } - - return ret; -} - -static int davinci_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - - if (prtd->ram_channel >= 0) - edma_stop(prtd->ram_channel); - if (prtd->asp_channel >= 0) - edma_stop(prtd->asp_channel); - if (prtd->asp_link[0] >= 0) - edma_unlink(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_unlink(prtd->asp_link[1]); - if (prtd->ram_link >= 0) - edma_unlink(prtd->ram_link); - - if (prtd->asp_link[0] >= 0) - edma_free_slot(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_free_slot(prtd->asp_link[1]); - if (prtd->asp_channel >= 0) - edma_free_channel(prtd->asp_channel); - if (prtd->ram_link >= 0) - edma_free_slot(prtd->ram_link); - if (prtd->ram_link2 >= 0) - edma_free_slot(prtd->ram_link2); - if (prtd->ram_channel >= 0) - edma_free_channel(prtd->ram_channel); - - kfree(prtd); - - return 0; -} - -static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); -} - -static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int davinci_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops davinci_pcm_ops = { - .open = davinci_pcm_open, - .close = davinci_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = davinci_pcm_hw_params, - .hw_free = davinci_pcm_hw_free, - .prepare = davinci_pcm_prepare, - .trigger = davinci_pcm_trigger, - .pointer = davinci_pcm_pointer, - .mmap = davinci_pcm_mmap, -}; - -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " - "size=%d\n", (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void davinci_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - struct snd_dma_buffer *iram_dma; - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - iram_dma = buf->private_data; - if (iram_dma) { - davinci_free_sram(substream, iram_dma); - kfree(iram_dma); - } - } -} - -static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, - pcm_hardware_playback.buffer_bytes_max); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, - pcm_hardware_capture.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver davinci_soc_platform = { - .ops = &davinci_pcm_ops, - .pcm_new = davinci_pcm_new, - .pcm_free = davinci_pcm_free, -}; - -int davinci_soc_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &davinci_soc_platform); -} -EXPORT_SYMBOL_GPL(davinci_soc_platform_register); - -MODULE_AUTHOR("Vladimir Barinov"); -MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h deleted file mode 100644 index 0fe2346a9aa2..000000000000 --- a/sound/soc/davinci/davinci-pcm.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> - * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _DAVINCI_PCM_H -#define _DAVINCI_PCM_H - -#include <linux/genalloc.h> -#include <linux/platform_data/davinci_asp.h> -#include <linux/platform_data/edma.h> - -struct davinci_pcm_dma_params { - int channel; /* sync dma channel ID */ - unsigned short acnt; - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned sram_size; - struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned char data_type; /* xfer data type */ - unsigned char convert_mono_stereo; - unsigned int fifo_level; -}; - -#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC) -int davinci_soc_platform_register(struct device *dev); -#else -static inline int davinci_soc_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_DAVINCI_SOC */ - -#endif diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 5bee04279ebe..fabd05f24aeb 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -33,8 +33,9 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" #define MOD_REG_BIT(val, mask, set) do { \ @@ -47,7 +48,8 @@ struct davinci_vcif_dev { struct davinci_vc *davinci_vc; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; }; static void davinci_vcif_start(struct snd_pcm_substream *substream) @@ -93,8 +95,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, { struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - struct davinci_pcm_dma_params *dma_params = - &davinci_vcif_dev->dma_params[substream->stream]; u32 w; /* Restart the codec before setup */ @@ -113,16 +113,12 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: - dma_params->data_type = 0; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | DAVINCI_VC_CTRL_WD_UNSIGNED, 1); break; case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_WD_BITS_8, 1); @@ -130,8 +126,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, DAVINCI_VC_CTRL_WD_UNSIGNED, 0); break; case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | @@ -142,8 +136,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dma_params->acnt = dma_params->data_type; - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); return 0; @@ -172,24 +164,25 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_vcif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; +static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_vcif_dai = { + .probe = davinci_vcif_dai_probe, .playback = { .channels_min = 1, .channels_max = 2, @@ -225,16 +218,16 @@ static int davinci_vcif_probe(struct platform_device *pdev) /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = - davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - davinci_vc->davinci_vcif.dma_tx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = + &davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + davinci_vc->davinci_vcif.dma_tx_addr; /* DMA rx params */ - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = - davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - davinci_vc->davinci_vcif.dma_rx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = + &davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + davinci_vc->davinci_vcif.dma_rx_addr; dev_set_drvdata(&pdev->dev, davinci_vcif_dev); @@ -245,7 +238,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c8e2f7..de438871040b 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2595611e8a6d..b9fabbf69db6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; for (i = 0; i < 255; i++) { - /* The bclk rate must be smaller than 1/5 sysclk rate */ - if (factor * (i + 1) < 5) - continue; - tmprate = freq * factor * (i + 2); if (baudclk_is_used) @@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (clkrate * 5 > clk_get_rate(ssi_private->clk)) + continue; + clkrate /= factor; afreq = clkrate / (i + 1); diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index f8cf10e16ce9..20e7400e2611 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) /* Headphone jack detection */ if (gpio_is_valid(data->jack_gpio)) { - ret = snd_soc_jack_new(rtd->codec, "Headphone", - SND_JACK_HEADPHONE | SND_JACK_BTN_0, - &headset_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack, NULL, 0); if (ret) return ret; diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index a958937ab405..0653aa83c927 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, + mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..c49a408fc7a6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -176,11 +176,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; if (gpio_is_valid(priv->gpio_hp_det)) { - snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, - &simple_card_hp_jack); - snd_soc_jack_add_pins(&simple_card_hp_jack, - ARRAY_SIZE(simple_card_hp_jack_pins), - simple_card_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &simple_card_hp_jack, + simple_card_hp_jack_pins, + ARRAY_SIZE(simple_card_hp_jack_pins)); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; @@ -189,11 +189,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(priv->gpio_mic_det)) { - snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, - &simple_card_mic_jack); - snd_soc_jack_add_pins(&simple_card_mic_jack, - ARRAY_SIZE(simple_card_mic_jack_pins), - simple_card_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &simple_card_mic_jack, + simple_card_mic_jack_pins, + ARRAY_SIZE(simple_card_mic_jack_pins)); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479ad..fc5542034b9b 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -80,15 +80,9 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret = 0; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); - - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&broadwell_headset, - ARRAY_SIZE(broadwell_headset_pins), - broadwell_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); if (ret) return ret; @@ -110,9 +104,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 9832afe7d22c..d8b1f038da1c 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,6 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_card *card = runtime->card; struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); struct snd_soc_jack *jack = &drv->jack; @@ -100,13 +99,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 59308629043e..3b262d01c1b3 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -113,9 +113,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9..012227997ed9 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -169,17 +169,17 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, - &ctx->hp_jack); + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "HP jack creation failed %d\n", ret); return ret; } - ret = snd_soc_jack_new(codec, "Mic Jack", - SND_JACK_MICROPHONE, - &ctx->mic_jack); + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); return ret; @@ -203,9 +203,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff016621583a..bc8dcacd5e6a 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -178,9 +178,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -217,7 +215,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -240,13 +238,13 @@ static struct snd_soc_dai_link cht_dailink[] = { .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, + .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, @@ -285,7 +283,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5672", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51a52aa..00fddd3f5dfb 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -56,9 +56,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 90b7a57713a0..49c09a0add79 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -228,10 +228,13 @@ static void mfld_jack_check(unsigned int intr_status) { struct mfld_jack_data jack_data; + if (!mfld_codec) + return; + jack_data.mfld_jack = &mfld_jack; jack_data.intr_id = intr_status; - sn95031_jack_detection(&jack_data); + sn95031_jack_detection(mfld_codec, &jack_data); /* TODO: add american headset detection post gpiolib support */ } @@ -240,8 +243,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - mfld_codec = runtime->codec; - /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -254,20 +255,15 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); if (ret_val) { pr_err("jack creation failed\n"); return ret_val; } - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } ret_val = snd_soc_jack_add_zones(&mfld_jack, ARRAY_SIZE(mfld_zones), mfld_zones); if (ret_val) { @@ -275,6 +271,8 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } + mfld_codec = runtime->codec; + /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc * to find if anything is there or not */ @@ -359,8 +357,6 @@ static irqreturn_t snd_mfld_jack_detection(int irq, void *data) { struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; mfld_jack_check(mc_drv_ctx->interrupt_status); return IRQ_HANDLED; diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2a..daecc58f28af 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7523cbef8780..2fbaf2c75d17 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -594,11 +594,13 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); @@ -665,6 +667,9 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) static int sst_soc_probe(struct snd_soc_platform *platform) { + struct sst_data *drv = dev_get_drvdata(platform->dev); + + drv->soc_card = platform->component.card; return sst_dsp_init_v2_dpcm(platform); } @@ -727,9 +732,64 @@ static int sst_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP + +static int sst_soc_prepare(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* suspend all pcms first */ + snd_soc_suspend(drv->soc_card->dev); + snd_soc_poweroff(drv->soc_card->dev); + + /* set the SSPs to idle */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } + } + + return 0; +} + +static void sst_soc_complete(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* restart SSPs */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + sst_handle_vb_timer(dai, true); + send_ssp_cmd(dai, dai->name, 1); + } + } + snd_soc_resume(drv->soc_card->dev); +} + +#else + +#define sst_soc_prepare NULL +#define sst_soc_complete NULL + +#endif + + +static const struct dev_pm_ops sst_platform_pm = { + .prepare = sst_soc_prepare, + .complete = sst_soc_complete, +}; + static struct platform_driver sst_platform_driver = { .driver = { .name = "sst-mfld-platform", + .pm = &sst_platform_pm, }, .probe = sst_platform_probe, .remove = sst_platform_remove, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 79c8d1246a8f..9094314be2b0 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -174,6 +174,7 @@ struct sst_data { struct sst_platform_data *pdata; struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; + struct snd_soc_card *soc_card; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e7..1a7eeec444b1 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx, spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - shim_regs->imrx = sst_shim_read64(shim, SST_IMRX), + shim_regs->imrx = sst_shim_read64(shim, SST_IMRX); + shim_regs->csr = sst_shim_read64(shim, SST_CSR); + spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } @@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx, */ spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); sst_shim_write64(shim, SST_IMRX, shim_regs->imrx), + sst_shim_write64(shim, SST_CSR, shim_regs->csr), spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } @@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ + + if (!acpi_disabled) + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); if (acpi_disabled) @@ -409,29 +416,142 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); return ret; } -static int intel_sst_runtime_resume(struct device *dev) +static int intel_sst_suspend(struct device *dev) { - int ret = 0; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save; + int i, ret = 0; - if (ctx->sst_state == SST_RESET) { - ret = sst_load_fw(ctx); - if (ret) { - dev_err(dev, "FW download fail %d\n", ret); - sst_set_fw_state_locked(ctx, SST_RESET); + /* check first if we are already in SW reset */ + if (ctx->sst_state == SST_RESET) + return 0; + + /* + * check if any stream is active and running + * they should already by suspend by soc_suspend + */ + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->status == STREAM_RUNNING) { + dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + return -EBUSY; } } + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + /* tell DSP we are suspending */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* save the memories */ + fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); + if (!fw_save) + return -ENOMEM; + fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + if (!fw_save->iram) { + ret = -ENOMEM; + goto iram; + } + fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + if (!fw_save->dram) { + ret = -ENOMEM; + goto dram; + } + fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + if (!fw_save->sram) { + ret = -ENOMEM; + goto sram; + } + + fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + if (!fw_save->ddr) { + ret = -ENOMEM; + goto ddr; + } + + memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base); + memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base); + memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE); + memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base); + + ctx->fw_save = fw_save; + ctx->ops->reset(ctx); + return 0; +ddr: + kfree(fw_save->sram); +sram: + kfree(fw_save->dram); +dram: + kfree(fw_save->iram); +iram: + kfree(fw_save); + return ret; +} + +static int intel_sst_resume(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save = ctx->fw_save; + int ret = 0; + struct sst_block *block; + + if (!fw_save) + return 0; + + sst_set_fw_state_locked(ctx, SST_FW_LOADING); + + /* we have to restore the memory saved */ + ctx->ops->reset(ctx); + + ctx->fw_save = NULL; + + memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base); + memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base); + memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); + memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); + + kfree(fw_save->sram); + kfree(fw_save->dram); + kfree(fw_save->iram); + kfree(fw_save->ddr); + kfree(fw_save); + + block = sst_create_block(ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + + /* start and wait for ack */ + ctx->ops->start(ctx); + ret = sst_wait_timeout(ctx, block); + if (ret) { + dev_err(ctx->dev, "fw download failed %d\n", ret); + /* FW download failed due to timeout */ + ret = -EBUSY; + + } else { + sst_set_fw_state_locked(ctx, SST_FW_RUNNING); + } + + sst_free_block(ctx, block); return ret; } const struct dev_pm_ops intel_sst_pm = { + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, - .runtime_resume = intel_sst_runtime_resume, }; EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 562bc483d6b7..3f493862e98d 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,6 +337,13 @@ struct sst_shim_regs64 { u64 csr2; }; +struct sst_fw_save { + void *iram; + void *dram; + void *sram; + void *ddr; +}; + /** * struct intel_sst_drv - driver ops * @@ -428,6 +435,8 @@ struct intel_sst_drv { * persistent till worker thread gets called */ char firmware_name[FW_NAME_SIZE]; + + struct sst_fw_save *fw_save; }; /* misc definitions */ @@ -544,4 +553,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, int sst_context_init(struct intel_sst_drv *ctx); void sst_context_cleanup(struct intel_sst_drv *ctx); void sst_configure_runtime_pm(struct intel_sst_drv *ctx); +void memcpy32_toio(void __iomem *dst, const void *src, int count); +void memcpy32_fromio(void *dst, const void __iomem *src, int count); + #endif diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5f75ef3cdd22..f0e4b99b3aeb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -138,12 +138,36 @@ int sst_get_stream(struct intel_sst_drv *ctx, static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); - - dev_dbg(ctx->dev, "state:%d", state); - if (state == true) - return pm_runtime_get_sync(dev); - else + int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif + + if (state == true) { + ret = pm_runtime_get_sync(dev); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); + if (ret < 0) { + dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); + return ret; + } + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + sst_set_fw_state_locked(ctx, SST_RESET); + ret = sst_pm_runtime_put(ctx); + } + } + } else { + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); + } + return ret; } /* @@ -572,6 +596,35 @@ static int sst_stream_drop(struct device *dev, int str_id) return sst_drop_stream(ctx, str_id); } +static int sst_stream_pause(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + return sst_pause_stream(ctx, str_id); +} + +static int sst_stream_resume(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + return sst_resume_stream(ctx, str_id); +} + static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) { int str_id = 0; @@ -633,6 +686,8 @@ static struct sst_ops pcm_ops = { .stream_init = sst_stream_init, .stream_start = sst_stream_start, .stream_drop = sst_stream_drop, + .stream_pause = sst_stream_pause, + .stream_pause_release = sst_stream_resume, .stream_read_tstamp = sst_read_timestamp, .send_byte_stream = sst_send_byte_stream, .close = sst_close_pcm_stream, diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 7888cd707853..e88907ae8b15 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,7 +39,15 @@ #include "sst.h" #include "../sst-dsp.h" -static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) +void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); +} + +void memcpy32_fromio(void *dst, const void __iomem *src, int count) { /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac55..e7c78b0406b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 706613077c15..16cc95fa4573 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -479,8 +479,8 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ - ret = snd_soc_jack_new(rtd->codec, "hook_switch", - SND_JACK_HEADSET, &ams_delta_hook_switch); + ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, + &ams_delta_hook_switch, NULL, 0); if (ret) dev_warn(card->dev, "Failed to allocate resources for hook switch, " diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index b9c65f1ad5a8..0843a68f277c 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -182,17 +182,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Headset jack detection only if it is supported */ if (priv->jack_detection) { - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - return ret; + return 0; } static const struct snd_soc_dapm_route dmic_audio_map[] = { diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index ccfb41c22e53..f7eb42aa3f38 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) return ret; card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->name = devm_kasprintf(dev, GFP_KERNEL, "HDMI %s", dev_name(ad->dssdev)); card->owner = THIS_MODULE; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index c7eb9dd67f60..fd99d89de6a8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4b..6bb623a2a4df 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -201,7 +219,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64)); + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); if (ret) return ret; @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fb1f6bb87cd4..3673ada43bfb 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -170,14 +170,10 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) if (priv->jack_detect > 0) { hs_jack_gpios[0].gpio = priv->jack_detect; - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &priv->hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&priv->hs_jack, - ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &priv->hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7f299357c2d2..c2ddf0fbfa28 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -311,9 +311,9 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) } /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); + err = snd_soc_card_jack_new(rtd->card, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack, NULL, 0); if (err) { dev_err(card->dev, "Failed to add AV Jack\n"); return err; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 73eb5ddf9753..9f8be7cd567e 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -126,17 +126,12 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = { */ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), - hs_jack_pin); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, + ARRAY_SIZE(hs_jack_pin)); if (err) return err; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 910336c5ebeb..c20bbc042425 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -75,17 +75,12 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (err) return err; diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 5001dbb9b257..1753c7d9e760 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -78,15 +78,12 @@ static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack, mic_jack_pins, + ARRAY_SIZE(mic_jack_pins)); /* headphone, microphone detection & headset short detection */ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 76ccb172d0a7..bcbfbe8303f7 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -143,13 +143,9 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Jack detection API stuff */ - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &hs_jack); - if (ret) - goto err; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) goto err; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3cebf6ca03df..0632a36852c8 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -206,7 +206,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -223,7 +223,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631 diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 59b044255b78..c72e9fb26658 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -162,13 +162,8 @@ static struct platform_device *s3c24xx_snd_device; static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 141519c21e21..31a820eb0ac3 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -260,12 +260,12 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | - SND_JACK_BTN_4 | SND_JACK_BTN_5, - &littlemill_headset); + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset, NULL, 0); if (ret) return ret; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 243dea7ba38f..5f156093101e 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -56,16 +56,10 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &lowland_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&lowland_headset, - ARRAY_SIZE(lowland_headset_pins), - lowland_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &lowland_headset, lowland_headset_pins, + ARRAY_SIZE(lowland_headset_pins)); if (ret) return ret; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 873f2cb4bebe..35e37c457f1f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -211,13 +211,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 8291d2a5f152..dfbe2db1c407 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -151,13 +151,10 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); /* Headphone jack detection */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), - smartq_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack, + smartq_jack_pins, + ARRAY_SIZE(smartq_jack_pins)); if (err) return err; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 5ec7c52282f2..2dcb988bdff2 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,16 +153,10 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) pr_err("Failed to request HP_SEL GPIO: %d\n", ret); gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &speyside_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_headset, - ARRAY_SIZE(speyside_headset_pins), - speyside_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset, speyside_headset_pins, + ARRAY_SIZE(speyside_headset_pins)); if (ret) return ret; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 9c80506527c4..85ccfb7188cb 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -179,15 +179,10 @@ static int tobermory_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &tobermory_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&tobermory_headset, - ARRAY_SIZE(tobermory_headset_pins), - tobermory_headset_pins); + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | + SND_JACK_BTN_0, &tobermory_headset, + tobermory_headset_pins, + ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1b53605f7154..110577c52317 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_probe; } + dev_set_drvdata(dev, priv); + /* * asoc register */ @@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev) goto exit_snd_soc; } - dev_set_drvdata(dev, priv); - pm_runtime_enable(dev); dev_info(dev, "probed\n"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb..5c0658d49609 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1561,6 +1561,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1616,6 +1620,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { if (card->dai_link[i].dai_fmt) snd_soc_runtime_set_dai_fmt(&card->rtd[i], @@ -3223,8 +3231,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3308,8 +3316,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4380dcc064a5..9f60c25c4568 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,30 +22,42 @@ #include <trace/events/asoc.h> /** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC codec + * snd_soc_card_jack_new - Create a new jack + * @card: ASoC card * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack * @jack: structure to use for the jack + * @pins: Array of jack pins to be added to the jack or NULL + * @num_pins: Number of elements in the @pins array * * Creates a new jack object. * * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack) +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins) { + int ret; + mutex_init(&jack->mutex); - jack->codec = codec; + jack->card = card; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); + ret = snd_jack_new(card->snd_card, id, type, &jack->jack); + if (ret) + return ret; + + if (num_pins) + return snd_soc_jack_add_pins(jack, num_pins, pins); + + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_jack_new); +EXPORT_SYMBOL_GPL(snd_soc_card_jack_new); /** * snd_soc_jack_report - Report the current status for a jack @@ -63,7 +75,6 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; unsigned int sync = 0; @@ -74,8 +85,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->codec; - dapm = &codec->dapm; + dapm = &jack->card->dapm; mutex_lock(&jack->mutex); @@ -175,12 +185,12 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!pins[i].pin) { - dev_err(jack->codec->dev, "ASoC: No name for pin %d\n", + dev_err(jack->card->dev, "ASoC: No name for pin %d\n", i); return -EINVAL; } if (!pins[i].mask) { - dev_err(jack->codec->dev, "ASoC: No mask for pin %d" + dev_err(jack->card->dev, "ASoC: No mask for pin %d" " (%s)\n", i, pins[i].pin); return -EINVAL; } @@ -260,7 +270,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->component.card->dev; + struct device *dev = gpio->jack->card->dev; trace_snd_soc_jack_irq(gpio->name); @@ -299,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!gpios[i].name) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: No name for gpio at index %d\n", i); ret = -EINVAL; goto undo; @@ -320,7 +330,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } else { /* legacy GPIO number */ if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Invalid gpio %d\n", gpios[i].gpio); ret = -EINVAL; @@ -350,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Failed to mark GPIO at index %d as wake source: %d\n", i, ret); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e7cb88..6e3781e88f9a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2511,6 +2511,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; pcm->private_data = rtd; diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 769aca2fc5f5..6dcd06a966c7 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -106,11 +106,10 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack); - snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins), - tegra_alc5632_hs_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b752..6760f0ebc133 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -141,16 +141,14 @@ static const struct snd_kcontrol_new tegra_max98090_controls[] = { static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_max98090_hp_jack); - snd_soc_jack_add_pins(&tegra_max98090_hp_jack, - ARRAY_SIZE(tegra_max98090_hp_jack_pins), - tegra_max98090_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack, + tegra_max98090_hp_jack_pins, + ARRAY_SIZE(tegra_max98090_hp_jack_pins)); tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, @@ -159,11 +157,11 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(machine->gpio_mic_det)) { - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_max98090_mic_jack); - snd_soc_jack_add_pins(&tegra_max98090_mic_jack, - ARRAY_SIZE(tegra_max98090_mic_jack_pins), - tegra_max98090_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack, + tegra_max98090_mic_jack_pins, + ARRAY_SIZE(tegra_max98090_mic_jack_pins)); tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index ed759a3076b8..773daecaa5e8 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -108,15 +108,11 @@ static const struct snd_kcontrol_new tegra_rt5640_controls[] = { static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_rt5640_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, - ARRAY_SIZE(tegra_rt5640_hp_jack_pins), - tegra_rt5640_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978a6e3a..68d8b67e79c1 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -146,10 +146,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_rt5677_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, - &tegra_rt5677_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack, + &tegra_rt5677_hp_jack_pins, 1); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -158,10 +157,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_rt5677_mic_jack); - snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, - &tegra_rt5677_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack, + &tegra_rt5677_mic_jack_pins, 1); if (gpio_is_valid(machine->gpio_mic_present)) { tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e52420dae2b4..4a95b70f0cf0 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,21 +177,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack, + tegra_wm8903_hp_jack_pins, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins)); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_wm8903_mic_jack); - snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, - ARRAY_SIZE(tegra_wm8903_mic_jack_pins), - tegra_wm8903_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack, + tegra_wm8903_mic_jack_pins, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); |