diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2021-07-23 09:58:23 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2021-07-23 09:58:23 -0700 |
commit | e7562a00c1f54116f5a058e7e3ddd500188f60b2 (patch) | |
tree | 1f280d9f6e61ee3363e0c4f56bdc43b8d05ecea5 /sound/soc | |
parent | 8baef6386baaefb776bdd09b5c7630cf057c51c6 (diff) | |
parent | b0084afde27fe8a504377dee65f55bc6aa776937 (diff) | |
download | linux-e7562a00c1f54116f5a058e7e3ddd500188f60b2.tar.bz2 |
Merge tag 'sound-5.14-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes, mostly covering device-specific
regressions and bugs over ASoC, HD-audio and USB-audio, while
the ALSA PCM core received a few additional fixes for the
possible (new and old) regressions"
* tag 'sound-5.14-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (29 commits)
ALSA: usb-audio: Add registration quirk for JBL Quantum headsets
ALSA: hda/hdmi: Add quirk to force pin connectivity on NUC10
ALSA: pcm: Fix mmap without buffer preallocation
ALSA: pcm: Fix mmap capability check
ALSA: hda: intel-dsp-cfg: add missing ElkhartLake PCI ID
ASoC: ti: j721e-evm: Check for not initialized parent_clk_id
ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup
ALSA: hda/realtek: Fix pop noise and 2 Front Mic issues on a machine
ALSA: hdmi: Expose all pins on MSI MS-7C94 board
ALSA: sb: Fix potential ABBA deadlock in CSP driver
ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend
ASoC: amd: reverse stop sequence for stoneyridge platform
ASoC: soc-pcm: add a flag to reverse the stop sequence
ASoC: codecs: wcd938x: setup irq during component bind
ASoC: dt-bindings: renesas: rsnd: Fix incorrect 'port' regex schema
ALSA: usb-audio: Add missing proc text entry for BESPOKEN type
ASoC: codecs: wcd938x: make sdw dependency explicit in Kconfig
ASoC: SOF: Intel: Update ADL descriptor to use ACPI power states
ASoC: rt5631: Fix regcache sync errors on resume
ALSA: pcm: Call substream ack() method upon compat mmap commit
...
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/amd/acp-da7219-max98357a.c | 5 | ||||
-rw-r--r-- | sound/soc/codecs/Kconfig | 8 | ||||
-rw-r--r-- | sound/soc/codecs/rt5631.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/rt5682.c | 8 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic31xx.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic31xx.h | 4 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic32x4.c | 27 | ||||
-rw-r--r-- | sound/soc/codecs/wcd938x.c | 18 | ||||
-rw-r--r-- | sound/soc/codecs/wm_adsp.c | 6 | ||||
-rw-r--r-- | sound/soc/intel/boards/sof_sdw_max98373.c | 81 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 22 | ||||
-rw-r--r-- | sound/soc/sof/intel/pci-tgl.c | 1 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_pcm.c | 30 | ||||
-rw-r--r-- | sound/soc/ti/j721e-evm.c | 18 |
14 files changed, 145 insertions, 87 deletions
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 84e3906abd4f..9449fb40a956 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ebae3f09435..a3b784ed4f70 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305 high-efficiency mono Class-D audio power amplifiers. config SND_SOC_SSM2518 - tristate + tristate "Analog Devices SSM2518 Class-D Amplifier" depends on I2C config SND_SOC_SSM2602 @@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X Qualcomm SoCs like SDM845. config SND_SOC_WCD938X + depends on SND_SOC_WCD938X_SDW tristate config SND_SOC_WCD938X_SDW @@ -1813,11 +1814,6 @@ config SND_SOC_ZL38060 which consists of a Digital Signal Processor (DSP), several Digital Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. -config SND_SOC_ZX_AUD96P22 - tristate "ZTE ZX AUD96P22 CODEC" - depends on I2C - select REGMAP_I2C - # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3000bc128b5b..38356ea2bd6e 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e4c91571abae..abcd6f483788 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 51870d50f419..b504d63385b3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, ret); return ret; } + regcache_cache_only(aic31xx->regmap, true); + aic31xx->dev = &i2c->dev; aic31xx->irq = i2c->irq; diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c63b717040ed..dcd8aeb45cb3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); /* -12dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); - -static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1); static const char * const lo_cm_text[] = { @@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { }; static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = { - SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_SINGLE_S8_TLV("PCM Playback Volume", + AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm), SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum), - SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0, - tlv_driver_gain), - SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 6, 0x01, 1), - SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + SOC_SINGLE_TLV("HP Driver Gain Volume", + AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1), - SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1, - 0, 0, 117, 1, tlv_spk_vol), - SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2, - 4, 5, 0, tlv_amp_vol), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", + TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), }; static const struct snd_kcontrol_new hp_output_mixer_controls[] = { diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 78b76eceff8f..2fcc97370be2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0); } - ret = wcd938x_irq_init(wcd938x, component->dev); - if (ret) { - dev_err(component->dev, "%s: IRQ init failed: %d\n", - __func__, ret); - return ret; - } - wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, WCD938X_IRQ_HPHR_PDM_WD_INT); wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, @@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev) return PTR_ERR(wcd938x->regmap); } + ret = wcd938x_irq_init(wcd938x, dev); + if (ret) { + dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); + return ret; + } + + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; + ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 37aa020f23f6..549d98241dae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -282,6 +282,7 @@ /* * HALO_CCM_CORE_CONTROL */ +#define HALO_CORE_RESET 0x00000200 #define HALO_CORE_EN 0x00000001 /* @@ -1213,7 +1214,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size); + ret = wm_coeff_read_ctrl(ctl, ctl->cache, size); if (!ret && copy_to_user(bytes, ctl->cache, size)) ret = -EFAULT; @@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp) { return regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, - HALO_CORE_EN, HALO_CORE_EN); + HALO_CORE_RESET | HALO_CORE_EN, + HALO_CORE_RESET | HALO_CORE_EN); } static void wm_halo_stop_core(struct wm_adsp *dsp) diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c index 0e7ed906b341..25daef910aee 100644 --- a/sound/soc/intel/boards/sof_sdw_max98373.c +++ b/sound/soc/intel/boards/sof_sdw_max98373.c @@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd) +static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + int j; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* enable max98373 first */ - ret = max_98373_trigger(substream, cmd); - if (ret < 0) - break; - - ret = sdw_trigger(substream, cmd); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = sdw_trigger(substream, cmd); - if (ret < 0) - break; - - ret = max_98373_trigger(substream, cmd); - break; - default: - ret = -EINVAL; - break; + /* set spk pin by playback only */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + for_each_rtd_codec_dais(rtd, j, codec_dai) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(cpu_dai->component); + char pin_name[16]; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + if (enable) + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + else + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + + if (!ret) + snd_soc_dapm_sync(dapm); } - return ret; + return 0; +} + +static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_prepare dai link prepare is called first */ + ret = sdw_prepare(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, true); +} + +static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_hw_free dai link free is called first */ + ret = sdw_hw_free(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, false); } static const struct snd_soc_ops max_98373_sdw_ops = { .startup = sdw_startup, - .prepare = sdw_prepare, - .trigger = max98373_sdw_trigger, - .hw_free = sdw_hw_free, + .prepare = mx8373_sdw_prepare, + .trigger = sdw_trigger, + .hw_free = mx8373_sdw_hw_free, .shutdown = sdw_shutdown, }; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 46513bb97904..d1c570ca21ea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1015,6 +1015,7 @@ out: static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1055,14 +1056,23 @@ start_err: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a00262184efa..d04ce84fe7cc 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = { static const struct sof_dev_desc adl_desc = { .machines = snd_soc_acpi_intel_adl_machines, .alt_machines = snd_soc_acpi_intel_adl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 573374b89b10..d3276b4595af 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(tegra_pcm_pointer); -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, +static int tegra_pcm_preallocate_dma_buffer(struct device *dev, 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->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL); + buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->private_data = NULL; buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; + buf->dev.dev = dev; buf->bytes = size; return 0; @@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) if (!buf->area) return; - dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr); + dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } -static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd, +static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd, size_t size) { - struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); if (ret < 0) return ret; if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size); if (ret) goto err; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size); if (ret) goto err_free_play; } @@ -284,7 +281,16 @@ err: int tegra_pcm_construct(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max); + struct device *dev = component->dev; + + /* + * Fallback for backwards-compatibility with older device trees that + * have the iommus property in the virtual, top-level "sound" node. + */ + if (!of_get_property(dev->of_node, "iommus", NULL)) + dev = rtd->card->snd_card->dev; + + return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max); } EXPORT_SYMBOL_GPL(tegra_pcm_construct); diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, |