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author | Takashi Iwai <tiwai@suse.de> | 2018-04-25 12:22:20 +0200 |
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committer | Takashi Iwai <tiwai@suse.de> | 2018-04-25 12:22:20 +0200 |
commit | 3a230f7d09dd0f0eec90bc94abc440ab740c86f1 (patch) | |
tree | 882e7922b62bf925872ef1beac993307df39d137 /sound | |
parent | 65811834ba56e9ed88117cf6c09880416c9951ab (diff) | |
parent | a8419a0cd98ddf628a9e38a92110af7cc650dde7 (diff) | |
download | linux-3a230f7d09dd0f0eec90bc94abc440ab740c86f1.tar.gz linux-3a230f7d09dd0f0eec90bc94abc440ab740c86f1.tar.bz2 linux-3a230f7d09dd0f0eec90bc94abc440ab740c86f1.zip |
Merge tag 'asoc-fix-4.17-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.17
A small batch of fixes collected since the merge window, none of which
are particularly large or remarkable. They've all been cooking in -next
for a while.
Diffstat (limited to 'sound')
-rw-r--r-- | sound/soc/amd/acp-da7219-max98357a.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/adau17x1.c | 26 | ||||
-rw-r--r-- | sound/soc/codecs/adau17x1.h | 3 | ||||
-rw-r--r-- | sound/soc/codecs/msm8916-wcd-analog.c | 9 | ||||
-rw-r--r-- | sound/soc/codecs/rt5514.c | 3 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_esai.c | 7 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 14 | ||||
-rw-r--r-- | sound/soc/intel/Kconfig | 22 | ||||
-rw-r--r-- | sound/soc/omap/omap-dmic.c | 14 | ||||
-rw-r--r-- | sound/soc/sh/rcar/core.c | 4 | ||||
-rw-r--r-- | sound/soc/soc-topology.c | 14 |
11 files changed, 85 insertions, 33 deletions
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c782e494..f41560ecbcd1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -43,7 +43,7 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -struct clk *da7219_dai_clk; +static struct clk *da7219_dai_clk; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 80c2a06285bb..12bf24c26818 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, } if (adau->sigmadsp) { - ret = adau17x1_setup_firmware(adau, params_rate(params)); + ret = adau17x1_setup_firmware(component, params_rate(params)); if (ret < 0) return ret; } @@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate) { int ret; - int dspsr; + int dspsr, dsp_run; + struct adau *adau = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr); if (ret) - return ret; + goto err; + + ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run); + if (ret) + goto err; regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0); ret = sigmadsp_setup(adau->sigmadsp, rate); if (ret) { regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); - return ret; + goto err; } regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr); + regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run); - return 0; +err: + snd_soc_dapm_mutex_unlock(dapm); + + return ret; } EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index a7b1cb770814..e6fe87beec07 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct adau *adau, unsigned int rate); +int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 12ee83d52405..b7cf7cce95fe 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1187,7 +1187,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, pm8916_mbhc_switch_irq_handler, + ret = devm_request_threaded_irq(dev, irq, NULL, + pm8916_mbhc_switch_irq_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "mbhc switch irq", priv); @@ -1201,7 +1202,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, mbhc_btn_press_irq_handler, + ret = devm_request_threaded_irq(dev, irq, NULL, + mbhc_btn_press_irq_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "mbhc btn press irq", priv); @@ -1214,7 +1216,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return irq; } - ret = devm_request_irq(dev, irq, mbhc_btn_release_irq_handler, + ret = devm_request_threaded_irq(dev, irq, NULL, + mbhc_btn_release_irq_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "mbhc btn release irq", priv); diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index e8a66b03faab..1570b91bf018 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -89,6 +89,7 @@ static const struct reg_default rt5514_reg[] = { {RT5514_PLL3_CALIB_CTRL5, 0x40220012}, {RT5514_DELAY_BUF_CTRL1, 0x7fff006a}, {RT5514_DELAY_BUF_CTRL3, 0x00000000}, + {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, @@ -181,6 +182,7 @@ static bool rt5514_readable_register(struct device *dev, unsigned int reg) case RT5514_PLL3_CALIB_CTRL5: case RT5514_DELAY_BUF_CTRL1: case RT5514_DELAY_BUF_CTRL3: + case RT5514_ASRC_IN_CTRL1: case RT5514_DOWNFILTER0_CTRL1: case RT5514_DOWNFILTER0_CTRL2: case RT5514_DOWNFILTER0_CTRL3: @@ -238,6 +240,7 @@ static bool rt5514_i2c_readable_register(struct device *dev, case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL5: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL1: case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL3: + case RT5514_DSP_MAPPING | RT5514_ASRC_IN_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL1: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL2: case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL3: diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40a700493f4c..da8fd98c7f51 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + /* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */ + if (ratio <= 256) { + pm = ratio; + fp = 1; + goto out; + } + /* Set the max fluctuation -- 0.1% of the max devisor */ savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0823b08923b5..89df2d9f63d7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -217,6 +217,7 @@ struct fsl_ssi_soc_data { * @dai_fmt: DAI configuration this device is currently used with * @streams: Mask of current active streams: BIT(TX) and BIT(RX) * @i2s_net: I2S and Network mode configurations of SCR register + * (this is the initial settings based on the DAI format) * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode @@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } if (!fsl_ssi_is_ac97(ssi)) { + /* + * Keep the ssi->i2s_net intact while having a local variable + * to override settings for special use cases. Otherwise, the + * ssi->i2s_net will lose the settings for regular use cases. + */ + u8 i2s_net = ssi->i2s_net; + /* Normal + Network mode to send 16-bit data in 32-bit frames */ if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; + i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ if (channels == 1) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL; + i2s_net = SSI_SCR_I2S_MODE_NORMAL; regmap_update_bits(regs, REG_SSI_SCR, - SSI_SCR_I2S_NET_MASK, ssi->i2s_net); + SSI_SCR_I2S_NET_MASK, i2s_net); } /* In synchronous mode, the SSI uses STCCR for capture */ diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ceb105cbd461..addac2a8e52a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 09db2aec12a3..b2f5d2fa354d 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream, static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, unsigned int freq) { - struct clk *parent_clk; + struct clk *parent_clk, *mux; char *parent_clk_name; int ret = 0; @@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, return -ENODEV; } + mux = clk_get_parent(dmic->fclk); + if (IS_ERR(mux)) { + dev_err(dmic->dev, "can't get fck mux parent\n"); + clk_put(parent_clk); + return -ENODEV; + } + mutex_lock(&dmic->mutex); if (dmic->active) { /* disable clock while reparenting */ pm_runtime_put_sync(dmic->dev); - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); pm_runtime_get_sync(dmic->dev); } else { - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); } mutex_unlock(&dmic->mutex); @@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, dmic->fclk_freq = freq; err_busy: + clk_put(mux); clk_put(parent_clk); return ret; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a76688a8ba9..94f081b93258 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev) return ret; } -static int rsnd_suspend(struct device *dev) +static int __maybe_unused rsnd_suspend(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); @@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev) return 0; } -static int rsnd_resume(struct device *dev) +static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fa27d0fca6dc..986b8b2f90fb 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -513,7 +513,7 @@ static void remove_widget(struct snd_soc_component *comp, */ if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) { /* enumerated widget mixer */ - for (i = 0; i < w->num_kcontrols; i++) { + for (i = 0; w->kcontrols != NULL && i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; struct soc_enum *se = (struct soc_enum *)kcontrol->private_value; @@ -530,7 +530,7 @@ static void remove_widget(struct snd_soc_component *comp, } } else { /* volume mixer or bytes controls */ - for (i = 0; i < w->num_kcontrols; i++) { + for (i = 0; w->kcontrols != NULL && i < w->num_kcontrols; i++) { struct snd_kcontrol *kcontrol = w->kcontrols[i]; if (dobj->widget.kcontrol_type @@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(se); goto err_se; + } kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( be->hdr.name, be->hdr.access); kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(sbe); goto err; + } kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; @@ -2576,7 +2580,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) /* match index */ if (dobj->index != index && - dobj->index != SND_SOC_TPLG_INDEX_ALL) + index != SND_SOC_TPLG_INDEX_ALL) continue; switch (dobj->type) { |