December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] ALSA: Clear PCM buffers more properly
by Takashi Iwai 18 Dec '20

18 Dec '20

Hi, here are two small fix patches for PCM buffer allocations. Basically those address two things that have been discussed in the recent thread [1]: * Clear the remaining bytes in PCM buffer pages that are exposed via mmap * Assure the page aligned allocations for genalloc This doesn't cover the cases where non-standard PCM buffer allocation is used. Those should handle the buffer clearance by themselves. Takashi [1] https://lore.kernel.org/r/05c824e5-0c33-4182-26fa-b116a42b10d6@metafoo.de === Takashi Iwai (2): ALSA: memalloc: Align buffer allocations in page size ALSA: pcm: Clear the full allocated memory at hw_params sound/core/memalloc.c | 1 + sound/core/pcm_native.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.26.2

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[PATCH] ALSA: usb-audio: Disable sample read check if firmware doesn't give back
by Takashi Iwai 18 Dec '20

18 Dec '20

Some buggy firmware don't give the current sample rate but leaves zero. Handle this case more gracefully without warning but just skip the current rate verification from the next time. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index e940dcee792b..31051f2be46d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; -- 2.26.2

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Re: [PATCH 06/18] arm64: dts: renesas: beacon: Configure Audio CODEC clocks
by Geert Uytterhoeven 18 Dec '20

18 Dec '20

Hi Adam, CC alsa-devel On Sun, Dec 13, 2020 at 7:38 PM Adam Ford <aford173(a)gmail.com> wrote: > With the newly added configurable clock options, the audio CODEC can > configure the mclk automatically. Add the reference to the versaclock. > Since the devices on I2C5 can communicate at 400KHz, let's also increase > that too > > Signed-off-by: Adam Ford <aford173(a)gmail.com> Thanks for your patch! > --- a/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > +++ b/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > @@ -424,13 +424,15 @@ &i2c0 { > > &i2c5 { > status = "okay"; > - clock-frequency = <100000>; > + clock-frequency = <400000>; > pinctrl-0 = <&i2c5_pins>; > pinctrl-names = "default"; > > codec: wm8962@1a { > compatible = "wlf,wm8962"; > reg = <0x1a>; > + clocks = <&versaclock6_bb 3>; > + clock-names = "mclk"; While the driver does get the (nameless) clock, the DT bindings lack any mention of a clocks property. It would be good to update the bindings. Note that arch/arm/boot/dts/imx6-logicpd-baseboard.dtsi and arch/arm64/boot/dts/freescale/imx8mm-beacon-baseboard.dtsi (both by your hand) use "xclk" instead of "mclk"? > DCVDD-supply = <&reg_audio>; > DBVDD-supply = <&reg_audio>; > AVDD-supply = <&reg_audio>; Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Move SDM845 out of Qualcomm folder and add C360
by GitHub pull_request - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf pull request #70 was opened from Srinivas-Kandagatla: With driver_name fix in sdm845 machine driver https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… /proc/asound/cards get populated correctly with proper driver name and card long name. With this in place we should be able to rearrange sdm845 directory to be inline with others and reuse some of the common code. Without this all the sdm845 based machines will get a generic ucm description which will not work, one of such instance is Lenovo C360 Laptop. This patchset also add support to Lenovo C360 support Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: meson: axg-tdmin: fix axg skew offset
by Jerome Brunet 18 Dec '20

18 Dec '20

The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); -- 2.29.2

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[PATCH] ASoC: meson: axg-tdm-interface: fix loopback
by Jerome Brunet 18 Dec '20

18 Dec '20

When the axg-tdm-interface was introduced, the backend DAI was marked as an endpoint when DPCM was walking the DAPM graph to find a its BE. It is no longer the case since this commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Because of this, when DPCM finds a BE it does everything it needs on the DAIs but it won't power up the widgets between the FE and the BE if there is no actual endpoint after the BE. On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE. It is only linked to the dummy codec since there no actual HW after it. >From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM decoder, which is a widget between the FE and BE is not powered up. >From the user perspective, everything seems fine but no data is produced. Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem. Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c8664ab80d45..87cac440b369 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = { -- 2.29.2

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[PATCH] ASoC: Intel: haswell: Add missing pm_ops
by Cezary Rojewski 18 Dec '20

18 Dec '20

haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705..c763bfeb1f38 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.17.1

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[PATCH] ASoC: rt711: mutex between calibration and power state changes
by shumingf＠realtek.com 18 Dec '20

18 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid calibration time-out, this patch adds the mutex between calibration and power state changes Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5771c02c3459..85f744184a60 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, unsigned int read_ll, read_rl; int i; + mutex_lock(&rt711->calibrate_mutex); + /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; @@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + mutex_unlock(&rt711->calibrate_mutex); return 0; } @@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + mutex_lock(&rt711->calibrate_mutex); regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); break; default: -- 2.29.0

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[PATCH v6 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Changes Since v5: -- Removed redundant initialization of map variable in lpass-platform.c Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++--------------- sound/soc/qcom/lpass-platform.c | 50 ++++++++++++++++++++++++++++------------- 2 files changed, 37 insertions(+), 33 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v3] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v3: - Allocate cache dynamically. v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 8 ++++++++ 4 files changed, 79 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..85f6865019d4 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -19,6 +19,12 @@ #include <sound/tlv.h> #include "max98373.h" +static const u32 max98373_i2c_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R2000_SW_RESET, 0x00}, {MAX98373_R2001_INT_RAW1, 0x00}, @@ -472,6 +478,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -509,6 +520,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c, { int ret = 0; int reg = 0; + int i; struct max98373_priv *max98373 = NULL; max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL); @@ -534,6 +546,14 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg); + max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_i2c_cache_reg[i]; + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..b8d471d79e93 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -23,6 +23,12 @@ struct sdw_stream_data { struct sdw_stream_runtime *sdw_stream; }; +static const u32 max98373_sdw_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R0040_SCP_INIT_STAT_1, 0x00}, {MAX98373_R0041_SCP_INIT_MASK_1, 0x00}, @@ -245,6 +251,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -757,6 +768,7 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) { struct max98373_priv *max98373; int ret; + int i; struct device *dev = &slave->dev; /* Allocate and assign private driver data structure */ @@ -768,6 +780,14 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg); + max98373->cache = devm_kcalloc(dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_sdw_cache_reg[i]; + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..71f5a5228f34 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,11 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +217,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

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