March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.4 08/24] ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe
by Sasha Levin 25 Mar '21

25 Mar '21

From: Benjamin Rood <benjaminjrood(a)gmail.com> [ Upstream commit f86f58e3594fb0ab1993d833d3b9a2496f3c928c ] According to the SGTL5000 datasheet [1], the DAP_AVC_CTRL register has the following bit field definitions: | BITS | FIELD | RW | RESET | DEFINITION | | 15 | RSVD | RO | 0x0 | Reserved | | 14 | RSVD | RW | 0x1 | Reserved | | 13:12 | MAX_GAIN | RW | 0x1 | Max Gain of AVC in expander mode | | 11:10 | RSVD | RO | 0x0 | Reserved | | 9:8 | LBI_RESP | RW | 0x1 | Integrator Response | | 7:6 | RSVD | RO | 0x0 | Reserved | | 5 | HARD_LMT_EN | RW | 0x0 | Enable hard limiter mode | | 4:1 | RSVD | RO | 0x0 | Reserved | | 0 | EN | RW | 0x0 | Enable/Disable AVC | The original default value written to the DAP_AVC_CTRL register during sgtl5000_i2c_probe() was 0x0510. This would incorrectly write values to bits 4 and 10, which are defined as RESERVED. It would also not set bits 12 and 14 to their correct RESET values of 0x1, and instead set them to 0x0. While the DAP_AVC module is effectively disabled because the EN bit is 0, this default value is still writing invalid values to registers that are marked as read-only and RESERVED as well as not setting bits 12 and 14 to their correct default values as defined by the datasheet. The correct value that should be written to the DAP_AVC_CTRL register is 0x5100, which configures the register bits to the default values defined by the datasheet, and prevents any writes to bits defined as 'read-only'. Generally speaking, it is best practice to NOT attempt to write values to registers/bits defined as RESERVED, as it generally produces unwanted/undefined behavior, or errors. Also, all credit for this patch should go to my colleague Dan MacDonald <dmacdonald(a)curbellmedical.com> for finding this error in the first place. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Benjamin Rood <benjaminjrood(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20210219183308.GA2117@ubuntu-dev Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f5b59305c957..8a1e485982d8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, -- 2.30.1

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[PATCH AUTOSEL 5.4 07/24] ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit eee51df776bd6cac10a76b2779a9fdee3f622b2b ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index c506c9305043..829cf552fe3e 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ -- 2.30.1

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[PATCH AUTOSEL 5.4 06/24] ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10
by Sasha Levin 25 Mar '21

25 Mar '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit cfa26ed1f9f885c2fd8f53ca492989d1e16d0199 ] The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB, not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace apps which translate the dB scale to a linear scale. With the logarithmic dB scale being of by a factor of 10 we loose all precision in the lower area of the range when apps translate things to a linear scale. E.g. the 0 dB default, which corresponds with a value of 47 of the 0 - 127 range for the control, would be shown as 0/100 in alsa-mixer. Since the centi-dB values used in the TLV struct cannot represent the 0.375 dB step size used by these controls, change the TLV definition for them to specify a min and max value instead of min + stepsize. Note this mirrors commit 3f31f7d9b540 ("ASoC: rt5670: Fix dac- and adc- vol-tlv values being off by a factor of 10") which made the exact same change to the rt5670 codec driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210226143817.84287-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 747ca248bf10..3bc63fbcb188 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ -- 2.30.1

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[PATCH v2] ASoC: core: Don't set platform name when of_node is set
by Daniel Baluta 25 Mar '21

25 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> A DAI link has 3 components: * CPU * platform * codec(s) A component is specified via: * name * of_node * dai_name In order to avoid confusion when building a sound card we disallow matching by both name and of_node (1). soc_check_tplg_fes allows overriding certain BE links by overriding BE link name. This doesn't work well if BE link was specified via DT, because we end up with a link with both name and of_node specified which is conflicting with (1). In order to fix this we need to: * override of_node if component was specified via DT * override name, otherwise. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 522bae73640a..7a1c011a7fe0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1648,7 +1648,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platforms->name = component->name; + + if (component->dev->of_node) + dai_link->platforms->of_node = component->dev->of_node; + else + dai_link->platforms->name = component->name; /* convert non BE into BE */ if (!dai_link->no_pcm) { -- 2.27.0

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[PATCH AUTOSEL 5.10 29/39] ASoC: rt711: add snd_soc_component remove callback
by Sasha Levin 25 Mar '21

25 Mar '21

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 899b12542b0897f92de9ba30944937c39ebb246d ] We do some IO operations in the snd_soc_component_set_jack callback function and snd_soc_component_set_jack() will be called when soc component is removed. However, we should not access SoundWire registers when the bus is suspended. So set regcache_cache_only(regmap, true) to avoid accessing in the soc component removal process. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20210316005254.29699-1-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt711.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index a9b1b4180c47..93d86f7558e0 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -895,6 +895,13 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } +static void rt711_remove(struct snd_soc_component *component) +{ + struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt711->regmap, true); +} + static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -905,6 +912,7 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, + .remove = rt711_remove, }; static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, -- 2.30.1

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[PATCH AUTOSEL 5.10 28/39] ASoC: rt5659: Update MCLK rate in set_sysclk()
by Sasha Levin 25 Mar '21

25 Mar '21

From: Sameer Pujar <spujar(a)nvidia.com> [ Upstream commit dbf54a9534350d6aebbb34f5c1c606b81a4f35dd ] Simple-card/audio-graph-card drivers do not handle MCLK clock when it is specified in the codec device node. The expectation here is that, the codec should actually own up the MCLK clock and do necessary setup in the driver. Suggested-by: Mark Brown <broonie(a)kernel.org> Suggested-by: Michael Walle <michael(a)walle.cc> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/1615829492-8972-3-git-send-email-spujar@nvidia.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5659.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 41e5917b16a5..91a4ef7f620c 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3426,12 +3426,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: -- 2.30.1

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[PATCH AUTOSEL 5.10 18/39] ASoC: cs42l42: Always wait at least 3ms after reset
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 19325cfea04446bc79b36bffd4978af15f46a00e ] This delay is part of the power-up sequence defined in the datasheet. A runtime_resume is a power-up so must also include the delay. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-6-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 3 ++- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5078ce79fad..4d82d24c7828 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1794,7 +1794,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1919,6 +1919,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9b017b76828a..866d7c873e3c 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -740,6 +740,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", -- 2.30.1

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[PATCH AUTOSEL 5.10 17/39] ASoC: cs42l42: Fix mixer volume control
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 72d904763ae6a8576e7ad034f9da4f0e3c44bf24 ] The minimum value is 0x3f (-63dB), which also is mute Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-4-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 4f9ad9547929..d5078ce79fad 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -401,7 +401,7 @@ static const struct regmap_config cs42l42_regmap = { }; static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -458,7 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, -- 2.30.1

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[PATCH AUTOSEL 5.10 16/39] ASoC: cs42l42: Fix channel width support
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit 2bdc4f5c6838f7c3feb4fe68e4edbeea158ec0a2 ] Remove the hard coded 32 bits width and replace with the correct width calculated by params_width. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-3-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 47 ++++++++++++++++++-------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index df0d5fec0287..4f9ad9547929 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -691,24 +691,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -824,14 +806,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -896,9 +893,9 @@ static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 1f0d67c95a9a..9b017b76828a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -757,7 +757,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.30.1

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[PATCH AUTOSEL 5.10 15/39] ASoC: cs42l42: Fix Bitclock polarity inversion
by Sasha Levin 25 Mar '21

25 Mar '21

From: Lucas Tanure <tanureal(a)opensource.cirrus.com> [ Upstream commit e793c965519b8b7f2fea51a48398405e2a501729 ] The driver was setting bit clock polarity opposite to intended polarity. Also simplify the code by grouping ADC and DAC clock configurations into a single field. Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210305173442.195740-2-tanureal@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 20 ++++++++------------ sound/soc/codecs/cs42l42.h | 11 ++++++----- 2 files changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 210fcbedf241..df0d5fec0287 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -797,27 +797,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9e3cc528dcff..1f0d67c95a9a 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -258,11 +258,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 -- 2.30.1

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