From da145172b236b1ac322fa81f6250aa59074eba68 Mon Sep 17 00:00:00 2001 From: "derek.fang" Date: Mon, 14 Sep 2020 16:57:18 +0800 Subject: [PATCH 01/22] ASoC: rt1015: Fix DC calibration on bypass boost mode Fix the DC calibration unsuccessful issue on rt1015 bypass boost mode. Signed-off-by: derek.fang Link: https://lore.kernel.org/r/1600073839-6762-1-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 48 +++++++++++++++++++++++++++------------ sound/soc/codecs/rt1015.h | 1 + 2 files changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 548f68649064..ba1b6b3822a5 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -484,6 +484,33 @@ static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } +static void rt1015_calibrate(struct rt1015_priv *rt1015) +{ + struct snd_soc_component *component = rt1015->component; + struct regmap *regmap = rt1015->regmap; + + snd_soc_dapm_mutex_lock(&component->dapm); + regcache_cache_bypass(regmap, true); + + regmap_write(regmap, RT1015_PWR1, 0xd7df); + regmap_write(regmap, RT1015_PWR4, 0x00b2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); + msleep(500); + regmap_write(regmap, RT1015_PWR1, 0x0); + + regcache_cache_bypass(regmap, false); + regcache_mark_dirty(regmap); + regcache_sync(regmap); + snd_soc_dapm_mutex_unlock(&component->dapm); +} + static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -494,20 +521,12 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, if (!rt1015->dac_is_used) { rt1015->bypass_boost = ucontrol->value.integer.value[0]; - if (rt1015->bypass_boost == RT1015_Bypass_Boost) { - snd_soc_component_write(component, - RT1015_PWR4, 0x00b2); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL8, 0x2008); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL9, 0x0140); - snd_soc_component_write(component, - RT1015_GAT_BOOST, 0x0efe); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000d); - msleep(500); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000e); + if (rt1015->bypass_boost == RT1015_Bypass_Boost && + !rt1015->cali_done) { + rt1015_calibrate(rt1015); + rt1015->cali_done = 1; + + regmap_write(rt1015->regmap, RT1015_MONO_DYNA_CTRL, 0x0010); } } else dev_err(component->dev, "DAC is being used!\n"); @@ -888,6 +907,7 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; + rt1015->cali_done = 0; snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); return 0; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 7bd159e8f958..4d11f5865725 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -389,6 +389,7 @@ struct rt1015_priv { int bypass_boost; int amp_ver; int dac_is_used; + int cali_done; }; #endif /* __RT1015_H__ */ From 8d9a14fc7371a18d54226bf2542f77b11e5a7101 Mon Sep 17 00:00:00 2001 From: "derek.fang" Date: Mon, 14 Sep 2020 16:57:19 +0800 Subject: [PATCH 02/22] ASoC: rt1015: Fix the failure to flush DAC data before playback Fix the failure to flush DAC data before playback. Signed-off-by: derek.fang Link: https://lore.kernel.org/r/1600073839-6762-2-git-send-email-derek.fang@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1015.c | 63 ++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt1015.h | 7 +++++ 2 files changed, 63 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index ba1b6b3822a5..25fe2ddedd54 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -534,6 +534,32 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, return 0; } +static void rt1015_flush_work(struct work_struct *work) +{ + struct rt1015_priv *rt1015 = container_of(work, struct rt1015_priv, + flush_work.work); + struct snd_soc_component *component = rt1015->component; + unsigned int val, i = 0, count = 20; + + while (i < count) { + usleep_range(1000, 1500); + dev_dbg(component->dev, "Flush DAC (retry:%u)\n", i); + regmap_read(rt1015->regmap, RT1015_CLK_DET, &val); + if (val & 0x800) + break; + i++; + } + + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); + + if (val & 0x800) + dev_dbg(component->dev, "Flush DAC completed.\n"); + else + dev_warn(component->dev, "Fail to flush DAC data.\n"); +} + static const struct snd_kcontrol_new rt1015_snd_controls[] = { SOC_SINGLE_TLV("DAC Playback Volume", RT1015_DAC1, RT1015_DAC_VOL_SFT, 127, 0, dac_vol_tlv), @@ -587,12 +613,7 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - if (rt1015->bypass_boost == RT1015_Bypass_Boost) { - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); - } + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); break; case SND_SOC_DAPM_POST_PMD: @@ -608,6 +629,8 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_SYS_RST1, 0x05f5); } rt1015->dac_is_used = 0; + + cancel_delayed_work_sync(&rt1015->flush_work); break; default: @@ -616,6 +639,24 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt1015->hw_config == RT1015_HW_28) + schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + break; + default: + break; + } + return 0; +} + static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT1015_PWR1, RT1015_PWR_LDO2_BIT, 0, NULL, 0), @@ -649,6 +690,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), }; @@ -667,7 +710,8 @@ static const struct snd_soc_dapm_route rt1015_dapm_routes[] = { { "DAC", NULL, "MIXERV" }, { "DAC", NULL, "SUMV" }, { "DAC", NULL, "VREFLV" }, - { "SPO", NULL, "DAC" }, + { "Amp Drv", NULL, "DAC" }, + { "SPO", NULL, "Amp Drv" }, }; static int rt1015_hw_params(struct snd_pcm_substream *substream, @@ -910,6 +954,8 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->cali_done = 0; snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); + INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); + return 0; } @@ -917,6 +963,7 @@ static void rt1015_remove(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + cancel_delayed_work_sync(&rt1015->flush_work); regmap_write(rt1015->regmap, RT1015_RESET, 0); } @@ -1042,6 +1089,8 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, return ret; } + rt1015->hw_config = (i2c->addr == 0x29) ? RT1015_HW_29 : RT1015_HW_28; + regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); if ((val != RT1015_DEVICE_ID_VAL) && (val != RT1015_DEVICE_ID_VAL2)) { dev_err(&i2c->dev, diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 4d11f5865725..d3fdd30aca6d 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -373,6 +373,11 @@ enum { RT1015_Bypass_Boost, }; +enum { + RT1015_HW_28 = 0, + RT1015_HW_29, +}; + struct rt1015_priv { struct snd_soc_component *component; struct regmap *regmap; @@ -390,6 +395,8 @@ struct rt1015_priv { int amp_ver; int dac_is_used; int cali_done; + int hw_config; + struct delayed_work flush_work; }; #endif /* __RT1015_H__ */ From 944c517b8c838832a166f1c89afbf8724f4a6b49 Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Mon, 14 Sep 2020 20:24:34 +0300 Subject: [PATCH 03/22] ASoC: fsl_audmix: make clock and output src write only "alsactl -f state.conf store/restore" sequence fails because setting "mixing clock source" and "output source" requires active TDM clock being started for configuration propagation. Make these two controls write only so that their values are not stored at "alsactl store". Signed-off-by: Viorel Suman Acked-by: Nicolin Chen Link: https://lore.kernel.org/r/1600104274-13110-1-git-send-email-viorel.suman@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index a447bafa00d2..7ad5925772e8 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -199,10 +199,18 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { /* FSL_AUDMIX_CTR controls */ - SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], - snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), - SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], - snd_soc_get_enum_double, fsl_audmix_put_out_src), + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mixing Clock Source", + .info = snd_soc_info_enum_double, + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, + .put = fsl_audmix_put_mix_clk_src, + .private_value = (unsigned long)&fsl_audmix_enum[0] }, + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output Source", + .info = snd_soc_info_enum_double, + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, + .put = fsl_audmix_put_out_src, + .private_value = (unsigned long)&fsl_audmix_enum[1] }, SOC_ENUM("Output Width", fsl_audmix_enum[2]), SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), From 6788fc1a66a0c1d1cec7a0f84f94b517eae8611c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 17 Sep 2020 13:39:12 +0300 Subject: [PATCH 04/22] ASoC: topology: disable size checks for bytes_ext controls if needed When CONFIG_SND_CTL_VALIDATION is set, accesses to extended bytes control generate spurious error messages when the size exceeds 512 bytes, such as [ 11.224223] sof_sdw sof_sdw: control 2:0:0:EQIIR5.0 eqiir_coef_5:0: invalid count 1024 In addition the error check returns -EINVAL which has the nasty side effect of preventing applications accessing controls from working, e.g. root@plb:~# alsamixer cannot load mixer controls: Invalid argument It's agreed that the control interface has been abused since 2014, but forcing a check should not prevent existing solutions from working. This patch skips the checks conditionally if CONFIG_SND_CTL_VALIDATION is set and the byte array provided by topology is > 512. This preserves the checks for all other cases. Fixes: 1a3232d2f61d2 ('ASoC: topology: Add support for TLV bytes controls') BugLink: https://github.com/thesofproject/linux/issues/2430 Reported-by: Takashi Iwai Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Jaska Uimonen Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20200917103912.2565907-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index cee998671318..e5ff9a561951 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -592,6 +592,17 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->info = snd_soc_bytes_info_ext; k->tlv.c = snd_soc_bytes_tlv_callback; + /* + * When a topology-based implementation abuses the + * control interface and uses bytes_ext controls of + * more than 512 bytes, we need to disable the size + * checks, otherwise accesses to such controls will + * return an -EINVAL error and prevent the card from + * being configured. + */ + if (IS_ENABLED(CONFIG_SND_CTL_VALIDATION) && sbe->max > 512) + k->access |= SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK; + ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { From 73154aca4a03a2ab4833fd36683feb884af06d4b Mon Sep 17 00:00:00 2001 From: Camel Guo Date: Tue, 8 Sep 2020 11:04:17 +0200 Subject: [PATCH 05/22] ASoC: tlv320adcx140: Fix digital gain range According to its datasheet, the digital gain should be -100 dB when CHx_DVOL is 1 and 27 dB when CHx_DVOL is 255. But with the current dig_vol_tlv, "Digital CHx Out Volume" shows 27.5 dB if CHx_DVOL is 255 and -95.5 dB if CHx_DVOL is 1. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Camel Guo Link: https://lore.kernel.org/r/20200908090417.16695-1-camel.guo@axis.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8efe20605f9b..c7c782d279d0 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -161,7 +161,7 @@ static const struct regmap_config adcx140_i2c_regmap = { }; /* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10050, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); From df16e2210454ca0b8a59caf364dd287fbb76a804 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Tue, 15 Sep 2020 14:06:01 -0500 Subject: [PATCH 06/22] ASoC: tlv320adcx140: Idle the device while writing registers It was observed that if the device was active and register writes were performed there were some unwanted behaviors particularly when writing the word length and some filter options. So when writing to the device the device should be placed in sleep mode and then exit sleep mode once the register update is complete. Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200915190606.1744-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 43 ++++++++++++++++++++++++-------- sound/soc/codecs/tlv320adcx140.h | 2 ++ 2 files changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index c7c782d279d0..5e14b9f8c84f 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -30,7 +30,7 @@ struct adcx140_priv { struct regmap *regmap; struct device *dev; - int micbias_vg; + bool micbias_vg; unsigned int dai_fmt; unsigned int tdm_delay; @@ -614,11 +614,26 @@ static int adcx140_reset(struct adcx140_priv *adcx140) return ret; } +static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) +{ + int pwr_ctrl = 0; + + if (power_state) + pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; + + if (adcx140->micbias_vg && power_state) + pwr_ctrl |= ADCX140_PWR_CFG_BIAS_PDZ; + + regmap_update_bits(adcx140->regmap, ADCX140_PWR_CFG, + ADCX140_PWR_CTRL_MSK, pwr_ctrl); +} + static int adcx140_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 data = 0; switch (params_width(params)) { @@ -640,9 +655,13 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_WORD_LEN_MSK, data); + adcx140_pwr_ctrl(adcx140, true); + return 0; } @@ -709,6 +728,8 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_FSYNCINV_BIT | ADCX140_BCLKINV_BIT | @@ -721,6 +742,7 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, ADCX140_TX_OFFSET_MASK, offset); + adcx140_pwr_ctrl(adcx140, true); return 0; } @@ -818,12 +840,11 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u32(adcx140->dev, "ti,mic-bias-source", &bias_source); - if (ret) + if (ret || bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { bias_source = ADCX140_MIC_BIAS_VAL_VREF; - - if (bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); - return -EINVAL; + adcx140->micbias_vg = false; + } else { + adcx140->micbias_vg = true; } ret = device_property_read_u32(adcx140->dev, "ti,vref-source", @@ -906,6 +927,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ADCX140_MIC_BIAS_VREF_MSK, bias_cfg); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); + + adcx140_pwr_ctrl(adcx140, true); out: return ret; } @@ -914,21 +937,19 @@ static int adcx140_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - int pwr_cfg = 0; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | - ADCX140_PWR_CFG_ADC_PDZ; + adcx140_pwr_ctrl(adcx140, true); break; case SND_SOC_BIAS_OFF: - pwr_cfg = 0x0; + adcx140_pwr_ctrl(adcx140, false); break; } - return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); + return 0; } static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index eedbc1d7221f..94c6d1fd2977 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -123,6 +123,7 @@ #define ADCX140_MIC_BIAS_VREF_1375V 2 #define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_PWR_CTRL_MSK GENMASK(7, 5) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) #define ADCX140_PWR_CFG_PLL_PDZ BIT(5) @@ -145,4 +146,5 @@ #define ADCX140_GPO_CFG_MAX 4 #define ADCX140_GPO_DRV_MAX 5 + #endif /* _TLV320ADCX140_ */ From 244ac15de75ca62ed7a09c7291b67aeead9e12ac Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Tue, 15 Sep 2020 14:06:02 -0500 Subject: [PATCH 07/22] ASoC: tlv320adcx140: Fix BCLK inversion for DSP modes Fix the BCLK inversion for DSP modes This is how it is defined by ASoC: * BCLK: * - "normal" polarity means signal is available at rising edge of BCLK * - "inverted" polarity means signal is available at falling edge of BCLK The adcx140 defines the BCLK edge based on coding type. The PCM (DSP_A/B) should drive on rising and sample on falling edge, so from ASoC pov, it is IB_NF. But from the codec pov if it is configured in DSP mode, then the BCLK should not be inverted, defaults to the coding standard. For i2s, it is NB_NF from ASoC pov (drive on falling, sample on rising). >From the codec's pov BCLK should not invert either, as this is the default for the coding. So, inversion must take the format into account: IB_NF + DSP_A/B == the codec bclk inversion should be disabled NB_NF + DSP_A/B == the codec bclk inversion should be enabled NB_NF + I2S == the codec bclk inversion should be disabled Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200915190606.1744-2-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320adcx140.c | 44 +++++++++++++++++--------------- 1 file changed, 23 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5e14b9f8c84f..936261e6c6d8 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -673,7 +673,7 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 iface_reg1 = 0; u8 iface_reg2 = 0; int offset = 0; - int width = adcx140->slot_width; + bool inverted_bclk = false; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -689,24 +689,6 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* signal polarity */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_IF: - iface_reg1 |= ADCX140_FSYNCINV_BIT; - break; - case SND_SOC_DAIFMT_IB_IF: - iface_reg1 |= ADCX140_BCLKINV_BIT | ADCX140_FSYNCINV_BIT; - break; - case SND_SOC_DAIFMT_IB_NF: - iface_reg1 |= ADCX140_BCLKINV_BIT; - break; - case SND_SOC_DAIFMT_NB_NF: - break; - default: - dev_err(component->dev, "Invalid DAI clock signal polarity\n"); - return -EINVAL; - } - /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -716,16 +698,36 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg1 |= ADCX140_LEFT_JUST_BIT; break; case SND_SOC_DAIFMT_DSP_A: - offset += (adcx140->tdm_delay * width + 1); + offset = 1; + inverted_bclk = true; break; case SND_SOC_DAIFMT_DSP_B: - offset += adcx140->tdm_delay * width; + inverted_bclk = true; break; default: dev_err(component->dev, "Invalid DAI interface format\n"); return -EINVAL; } + /* signal polarity */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + case SND_SOC_DAIFMT_IB_IF: + inverted_bclk = !inverted_bclk; + break; + case SND_SOC_DAIFMT_NB_IF: + iface_reg1 |= ADCX140_FSYNCINV_BIT; + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + dev_err(component->dev, "Invalid DAI clock signal polarity\n"); + return -EINVAL; + } + + if (inverted_bclk) + iface_reg1 |= ADCX140_BCLKINV_BIT; + adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; adcx140_pwr_ctrl(adcx140, false); From 5f2df2a4583b0d7b85054f0c1820f11a01936d35 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 21 Sep 2020 17:42:44 +0800 Subject: [PATCH 08/22] ASoC: rt700: wait for the delayed work to finish when the system suspends To avoid the IO error, we need to cancel the delayed work and wait for it to finish. Signed-off-by: Shuming Fan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200921094244.31869-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700-sdw.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 1d24bf040718..af6e17e457b4 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -490,6 +490,9 @@ static int __maybe_unused rt700_dev_suspend(struct device *dev) if (!rt700->hw_init) return 0; + cancel_delayed_work_sync(&rt700->jack_detect_work); + cancel_delayed_work_sync(&rt700->jack_btn_check_work); + regcache_cache_only(rt700->regmap, true); return 0; From b0bcbe615756d5923eec4e95996e4041627e5741 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 18 Sep 2020 14:05:40 -0500 Subject: [PATCH 09/22] ASoC: tas2770: Fix calling reset in probe tas2770_reset is called during i2c probe. The reset calls the snd_soc_component_write which depends on the tas2770->component being available. The component pointer is not set until codec_probe so move the reset to the codec_probe after the pointer is set. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200918190548.12598-1-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index c09851834395..03d7ad1885b8 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -575,6 +575,8 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; + tas2770_reset(tas2770); + return 0; } @@ -771,8 +773,6 @@ static int tas2770_i2c_probe(struct i2c_client *client, tas2770->channel_size = 0; tas2770->slot_width = 0; - tas2770_reset(tas2770); - result = tas2770_register_codec(tas2770); if (result) dev_err(tas2770->dev, "Register codec failed.\n"); From 4272caf34aa4699eca8e6e7f4a8e5ea2bde275c9 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 18 Sep 2020 14:05:41 -0500 Subject: [PATCH 10/22] ASoC: tas2770: Add missing bias level power states Add the BIAS_STANDBY and BIAS_PREPARE to the set_bias_level or else the driver will return -EINVAL which is not correct as they are valid states. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200918190548.12598-2-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 03d7ad1885b8..7c6f61946ab3 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -57,7 +57,12 @@ static int tas2770_set_bias_level(struct snd_soc_component *component, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); break; - + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_MUTE); + break; case SND_SOC_BIAS_OFF: snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, From 4b8ab8a7761fe2ba1c4e741703a848cb8f390f79 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 18 Sep 2020 14:05:43 -0500 Subject: [PATCH 11/22] ASoC: tas2770: Fix required DT properties in the code The devicetree binding indicates that the ti,asi-format, ti,imon-slot-no and ti,vmon-slot-no are not required but the driver requires them or it fails to probe. Honor the binding and allow these entries to be optional and set the corresponding values to the default values for each as defined in the data sheet. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200918190548.12598-4-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 7c6f61946ab3..bdfdad5f4f64 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -708,29 +708,28 @@ static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *tas2770) rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", &tas2770->asi_format); if (rc) { - dev_err(tas2770->dev, "Looking up %s property failed %d\n", - "ti,asi-format", rc); - goto end; + dev_info(tas2770->dev, "Property %s is missing setting default slot\n", + "ti,asi-format"); + tas2770->asi_format = 0; } rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", &tas2770->i_sense_slot); if (rc) { - dev_err(tas2770->dev, "Looking up %s property failed %d\n", - "ti,imon-slot-no", rc); - goto end; + dev_info(tas2770->dev, "Property %s is missing setting default slot\n", + "ti,imon-slot-no"); + tas2770->i_sense_slot = 0; } rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", &tas2770->v_sense_slot); if (rc) { - dev_err(tas2770->dev, "Looking up %s property failed %d\n", - "ti,vmon-slot-no", rc); - goto end; + dev_info(tas2770->dev, "Property %s is missing setting default slot\n", + "ti,vmon-slot-no"); + tas2770->v_sense_slot = 2; } -end: - return rc; + return 0; } static int tas2770_i2c_probe(struct i2c_client *client, From cadab0aefcbadf488b16caf2770430e69f4d7839 Mon Sep 17 00:00:00 2001 From: Dan Murphy Date: Fri, 18 Sep 2020 14:05:46 -0500 Subject: [PATCH 12/22] ASoC: tas2770: Fix error handling with update_bits snd_soc_update_bits returns a 1 when the bit was successfully updated, returns a 0 is no update was needed and a negative if the call failed. The code is currently failing the case of a successful update by just checking for a non-zero number. Modify these checks and return the error code only if there is a negative. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy Link: https://lore.kernel.org/r/20200918190548.12598-7-dmurphy@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 52 ++++++++++++++++++-------------------- 1 file changed, 24 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index bdfdad5f4f64..15cdd8b11a67 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -140,23 +140,18 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_MUTE); - if (ret) - goto end; break; case SND_SOC_DAPM_PRE_PMD: ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) - goto end; break; default: dev_err(tas2770->dev, "Not supported evevt\n"); return -EINVAL; } -end: if (ret < 0) return ret; @@ -248,6 +243,9 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) return -EINVAL; } + if (ret < 0) + return ret; + tas2770->channel_size = bitwidth; ret = snd_soc_component_update_bits(component, @@ -256,16 +254,15 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) TAS2770_TDM_CFG_REG5_50_MASK, TAS2770_TDM_CFG_REG5_VSNS_ENABLE | tas2770->v_sense_slot); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG6, TAS2770_TDM_CFG_REG6_ISNS_MASK | TAS2770_TDM_CFG_REG6_50_MASK, TAS2770_TDM_CFG_REG6_ISNS_ENABLE | tas2770->i_sense_slot); - -end: if (ret < 0) return ret; @@ -283,36 +280,35 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_48KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); - if (ret) - goto end; break; case 44100: ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); - if (ret) - goto end; break; case 96000: ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_48KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, @@ -323,8 +319,9 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, @@ -335,22 +332,22 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_48KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); - if (ret) - goto end; break; case 17640: ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_SMP_MASK, TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); - if (ret) - goto end; + if (ret < 0) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG0, TAS2770_TDM_CFG_REG0_31_MASK, @@ -360,7 +357,6 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) ret = -EINVAL; } -end: if (ret < 0) return ret; From 501ef013390b774e8e61000a78d1d640d6c3411d Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 21 Sep 2020 17:43:08 +0800 Subject: [PATCH 13/22] ASoC: rt711: wait for the delayed work to finish when the system suspends To avoid the IO error, we need to cancel the delayed work and wait for it to finish. Signed-off-by: Shuming Fan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20200921094308.31921-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 7efff130a638..9eabd30521c7 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -491,6 +491,10 @@ static int __maybe_unused rt711_dev_suspend(struct device *dev) if (!rt711->hw_init) return 0; + cancel_delayed_work_sync(&rt711->jack_detect_work); + cancel_delayed_work_sync(&rt711->jack_btn_check_work); + cancel_work_sync(&rt711->calibration_work); + regcache_cache_only(rt711->regmap, true); return 0; From 5b4458ebb4c8007dae7eaeb88cb52b2bb4879894 Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Fri, 11 Sep 2020 19:31:38 +0200 Subject: [PATCH 14/22] ASoC: tlv320aic32x4: Ensure a minimum delay before clock stabilization As indicated in the datasheet, a 10ms delay must be observed after programming the divisors. The lack of delay prevents the codec to work properly and the playback appears extremely slow and totally un-audible on a custom sama5 based board. Signed-off-by: Miquel Raynal Link: https://lore.kernel.org/r/20200911173140.29984-2-miquel.raynal@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-clk.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 156c153c12ab..2f78e6820c75 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -230,7 +230,14 @@ static int clk_aic32x4_pll_set_rate(struct clk_hw *hw, if (ret < 0) return -EINVAL; - return clk_aic32x4_pll_set_muldiv(pll, &settings); + ret = clk_aic32x4_pll_set_muldiv(pll, &settings); + if (ret) + return ret; + + /* 10ms is the delay to wait before the clocks are stable */ + msleep(10); + + return 0; } static int clk_aic32x4_pll_set_parent(struct clk_hw *hw, u8 index) From 40b37136287ba6b34aa2f1f6123f3d6d205dc2f0 Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Fri, 11 Sep 2020 19:31:39 +0200 Subject: [PATCH 15/22] ASoC: tlv320aic32x4: Fix bdiv clock rate derivation Current code expects a single channel to be always used. Fix this situation by forwarding the number of channels used. Then fix the derivation of the bdiv clock rate. Fixes: 96c3bb00239d ("ASoC: tlv320aic32x4: Dynamically Determine Clocking") Suggested-by: Alexandre Belloni Signed-off-by: Miquel Raynal Link: https://lore.kernel.org/r/20200911173140.29984-3-miquel.raynal@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 467802875c13..2e2d8e463655 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -665,7 +665,7 @@ static int aic32x4_set_processing_blocks(struct snd_soc_component *component, } static int aic32x4_setup_clocks(struct snd_soc_component *component, - unsigned int sample_rate) + unsigned int sample_rate, unsigned int channels) { u8 aosr; u16 dosr; @@ -753,7 +753,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, dosr); clk_set_rate(clocks[5].clk, - sample_rate * 32); + sample_rate * 32 * + channels); + return 0; } } @@ -775,7 +777,8 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, u8 iface1_reg = 0; u8 dacsetup_reg = 0; - aic32x4_setup_clocks(component, params_rate(params)); + aic32x4_setup_clocks(component, params_rate(params), + params_channels(params)); switch (params_width(params)) { case 16: From ec96690de82cee2cb028c07b1e72cb4a446ad03a Mon Sep 17 00:00:00 2001 From: Miquel Raynal Date: Fri, 11 Sep 2020 19:31:40 +0200 Subject: [PATCH 16/22] ASoC: tlv320aic32x4: Enable fast charge At power-up the analog circuits may take up to one full second before being charged with the default configuration. Using the analog blocks before they are ready generates a *very* crappy sound. Enable the fast charge feature, which will require a bit more power than normal charge but will definitely speed up the starting operation by shrinking this delay to up to 40 ms. Signed-off-by: Miquel Raynal Link: https://lore.kernel.org/r/20200911173140.29984-4-miquel.raynal@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++++++ sound/soc/codecs/tlv320aic32x4.h | 7 +++++++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2e2d8e463655..ea8cd4487536 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1013,6 +1013,14 @@ static int aic32x4_component_probe(struct snd_soc_component *component) AIC32X4_LADC_EN | AIC32X4_RADC_EN); snd_soc_component_write(component, AIC32X4_ADCSETUP, tmp_reg); + /* + * Enable the fast charging feature and ensure the needed 40ms ellapsed + * before using the analog circuits. + */ + snd_soc_component_write(component, AIC32X4_REFPOWERUP, + AIC32X4_REFPOWERUP_40MS); + msleep(40); + return 0; } diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 38f47704bb75..7550122e9f8a 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -96,6 +96,7 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_FLOATINGINPUT AIC32X4_REG(1, 58) #define AIC32X4_LMICPGAVOL AIC32X4_REG(1, 59) #define AIC32X4_RMICPGAVOL AIC32X4_REG(1, 60) +#define AIC32X4_REFPOWERUP AIC32X4_REG(1, 123) /* Bits, masks, and shifts */ @@ -205,6 +206,12 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_RMICPGANIN_IN1L_10K 0x10 #define AIC32X4_RMICPGANIN_CM1R_10K 0x40 +/* AIC32X4_REFPOWERUP */ +#define AIC32X4_REFPOWERUP_SLOW 0x04 +#define AIC32X4_REFPOWERUP_40MS 0x05 +#define AIC32X4_REFPOWERUP_80MS 0x06 +#define AIC32X4_REFPOWERUP_120MS 0x07 + /* Common mask and enable for all of the dividers */ #define AIC32X4_DIVEN BIT(7) #define AIC32X4_DIV_MASK GENMASK(6, 0) From 64952377548517a14d0a6521856dde9fd8356c90 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 25 Sep 2020 10:18:29 +0100 Subject: [PATCH 17/22] ASoC: cs47l15: Fix EPOUT->HPOUT1 Mono Mux routing EPOUT is always mono so should have a permanent routing through the HPOUT1 Mono Mux. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20200925091830.7675-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l15.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index a591e7457d11..254f9d96e766 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1089,6 +1089,7 @@ static const struct snd_soc_dapm_route cs47l15_dapm_routes[] = { { "HPOUT1 Demux", NULL, "OUT1R" }, { "OUT1R", NULL, "HPOUT1 Mono Mux" }, + { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, { "HPOUTL", "HPOUT", "HPOUT1 Demux" }, { "HPOUTR", "HPOUT", "HPOUT1 Demux" }, @@ -1268,7 +1269,6 @@ static irqreturn_t cs47l15_adsp2_irq(int irq, void *data) static const struct snd_soc_dapm_route cs47l15_mono_routes[] = { { "HPOUT1 Mono Mux", "HPOUT", "OUT1L" }, - { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, }; static int cs47l15_component_probe(struct snd_soc_component *component) From b03acae25e4a4ad600e91a8737c7eff2efe7d66a Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 25 Sep 2020 10:18:30 +0100 Subject: [PATCH 18/22] ASoC: cs47l35: Fix EPOUT->HPOUT1 Mono Mux routing EPOUT is always mono so should have a permanent routing through the HPOUT1 Mono Mux. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20200925091830.7675-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 7f5dd01f40c9..e967609da8a3 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1305,6 +1305,7 @@ static const struct snd_soc_dapm_route cs47l35_dapm_routes[] = { { "SPKOUTP", NULL, "OUT4L" }, { "OUT1R", NULL, "HPOUT1 Mono Mux" }, + { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, { "HPOUTL", "HPOUT", "HPOUT1 Demux" }, { "HPOUTR", "HPOUT", "HPOUT1 Demux" }, @@ -1550,7 +1551,6 @@ static irqreturn_t cs47l35_adsp2_irq(int irq, void *data) static const struct snd_soc_dapm_route cs47l35_mono_routes[] = { { "HPOUT1 Mono Mux", "HPOUT", "OUT1L" }, - { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, }; static int cs47l35_component_probe(struct snd_soc_component *component) From 22a16145af824f91014d07f8664114859900b9e6 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Fri, 18 Sep 2020 18:26:00 +0800 Subject: [PATCH 19/22] ASoC: fsl_sai: Instantiate snd_soc_dai_driver Instantiate snd_soc_dai_driver for independent symmetric control. Otherwise the symmetric setting may be overwritten by other instance. Fixes: 08fdf65e37d5 ("ASoC: fsl_sai: Add asynchronous mode support") Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1600424760-32071-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 19 +++++++++++-------- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdff739924e2..2ea354dd5434 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -694,7 +694,7 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static struct snd_soc_dai_driver fsl_sai_dai = { +static struct snd_soc_dai_driver fsl_sai_dai_template = { .probe = fsl_sai_dai_probe, .playback = { .stream_name = "CPU-Playback", @@ -966,12 +966,15 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + memcpy(&sai->cpu_dai_drv, &fsl_sai_dai_template, + sizeof(fsl_sai_dai_template)); + /* Sync Tx with Rx as default by following old DT binding */ sai->synchronous[RX] = true; sai->synchronous[TX] = false; - fsl_sai_dai.symmetric_rates = 1; - fsl_sai_dai.symmetric_channels = 1; - fsl_sai_dai.symmetric_samplebits = 1; + sai->cpu_dai_drv.symmetric_rates = 1; + sai->cpu_dai_drv.symmetric_channels = 1; + sai->cpu_dai_drv.symmetric_samplebits = 1; if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && of_find_property(np, "fsl,sai-asynchronous", NULL)) { @@ -988,9 +991,9 @@ static int fsl_sai_probe(struct platform_device *pdev) /* Discard all settings for asynchronous mode */ sai->synchronous[RX] = false; sai->synchronous[TX] = false; - fsl_sai_dai.symmetric_rates = 0; - fsl_sai_dai.symmetric_channels = 0; - fsl_sai_dai.symmetric_samplebits = 0; + sai->cpu_dai_drv.symmetric_rates = 0; + sai->cpu_dai_drv.symmetric_channels = 0; + sai->cpu_dai_drv.symmetric_samplebits = 0; } if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && @@ -1020,7 +1023,7 @@ static int fsl_sai_probe(struct platform_device *pdev) regcache_cache_only(sai->regmap, true); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &fsl_sai_dai, 1); + &sai->cpu_dai_drv, 1); if (ret) goto err_pm_disable; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..677ecfc1ec68 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -180,6 +180,7 @@ struct fsl_sai { unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; + struct snd_soc_dai_driver cpu_dai_drv; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; From 6bf28e8a05fda0547658fd51d0acc83dcac6c703 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 25 Sep 2020 11:59:08 +0100 Subject: [PATCH 20/22] MAINTAINERS: Update Cirrus Logic Codecs maintainers Add patches@opensource.cirrus.com to the main group of Cirrus Logic ASoC codec drivers. Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20200925105908.20640-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index f0068bceeb61..113eb26099e5 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4154,6 +4154,7 @@ CIRRUS LOGIC AUDIO CODEC DRIVERS M: James Schulman M: David Rhodes L: alsa-devel@alsa-project.org (moderated for non-subscribers) +L: patches@opensource.cirrus.com S: Maintained F: sound/soc/codecs/cs* From 20441614d89867142060d3bcd79cc111d8ba7a8e Mon Sep 17 00:00:00 2001 From: Adam Brickman Date: Thu, 1 Oct 2020 16:24:25 +0100 Subject: [PATCH 21/22] ASoC: wm_adsp: Pass full name to snd_ctl_notify A call to wm_adsp_write_ctl() could cause a kernel crash if it does not retrieve a valid kcontrol from snd_soc_card_get_kcontrol(). This can happen due to a missing control name prefix. Then, snd_ctl_notify() crashes when it tries to use the id field. Modified wm_adsp_write_ctl() to incorporate the name_prefix (if applicable) such that it is able to retrieve a valid id field from the kcontrol once the platform has booted. Fixes: eb65ccdb0836 ("ASoC: wm_adsp: Expose mixer control API") Signed-off-by: Adam Brickman Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20201001152425.8590-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 410cca57da52..344bd2c33bea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2049,6 +2049,7 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, { struct wm_coeff_ctl *ctl; struct snd_kcontrol *kcontrol; + char ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ret; ctl = wm_adsp_get_ctl(dsp, name, type, alg); @@ -2059,8 +2060,25 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, return -EINVAL; ret = wm_coeff_write_ctrl(ctl, buf, len); + if (ret) + return ret; + + if (ctl->flags & WMFW_CTL_FLAG_SYS) + return 0; + + if (dsp->component->name_prefix) + snprintf(ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s", + dsp->component->name_prefix, ctl->name); + else + snprintf(ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s", + ctl->name); + + kcontrol = snd_soc_card_get_kcontrol(dsp->component->card, ctl_name); + if (!kcontrol) { + adsp_err(dsp, "Can't find kcontrol %s\n", ctl_name); + return -EINVAL; + } - kcontrol = snd_soc_card_get_kcontrol(dsp->component->card, ctl->name); snd_ctl_notify(dsp->component->card->snd_card, SNDRV_CTL_EVENT_MASK_VALUE, &kcontrol->id); From ebb11d1d9fe2d6b4a47755f7f09b2b631046e308 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 6 Oct 2020 18:12:52 +0800 Subject: [PATCH 22/22] ASoC: mediatek: mt8183-da7219: fix wrong ops for I2S3 DA7219 uses I2S2 and I2S3 for input and output respectively. Commit 9e30251fb22e ("ASoC: mediatek: mt8183-da7219: support machine driver with rt1015") introduces a bug that: - If using I2S2 solely, MCLK to DA7219 is 256FS. - If using I2S3 solely, MCLK to DA7219 is 128FS. - If using I2S3 first and then I2S2, the MCLK changes from 128FS to 256FS. As a result, no sound output to the headset. Also no sound input from the headset microphone. Both I2S2 and I2S3 should set MCLK to 256FS. Fixes the wrong ops for I2S3. Fixes: 9e30251fb22e ("ASoC: mediatek: mt8183-da7219: support machine driver with rt1015") Signed-off-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20201006101252.1890385-1-tzungbi@google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 06d0a4f80fc1..a6c690c5308d 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -673,7 +673,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) if (card == &mt8183_da7219_max98357_card) { dai_link->be_hw_params_fixup = mt8183_i2s_hw_params_fixup; - dai_link->ops = &mt8183_mt6358_i2s_ops; + dai_link->ops = &mt8183_da7219_i2s_ops; dai_link->cpus = i2s3_max98357a_cpus; dai_link->num_cpus = ARRAY_SIZE(i2s3_max98357a_cpus);