September 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: soc: codecs: tlv320aic3x: Fix codec gpio-reset
by Benjamin Hahn 30 Sep '24

30 Sep '24

The TLV320AIC3007 requires a hardware reset after power-up for proper operation. After all power supplies are at their specified values, the RESET pin must be driven low for at least 10 ns. Even though the datasheet sais min 10ns, I needed more than 10ns when testing this out. 15ns worked for me. Signed-off-by: Benjamin Hahn <B.Hahn(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 56e795a00e22..d002fc0b99b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1367,8 +1367,11 @@ static int aic3x_regulator_event(struct notifier_block *nb, * Put codec to reset and require cache sync as at least one * of the supplies was disabled */ - if (aic3x->gpio_reset) + if (aic3x->gpio_reset) { gpiod_set_value(aic3x->gpio_reset, 1); + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + } regcache_mark_dirty(aic3x->regmap); } @@ -1813,6 +1816,10 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver gpiod_set_consumer_name(aic3x->gpio_reset, "tlv320aic3x reset"); + /* CODEC datasheet says minimum of 10ns */ + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; --- base-commit: ad46e8f95e931e113cb98253daf6d443ac244cde change-id: 20240930-wip-bhahn-tlv320aic3x_fix_reset-cca918a4889d Best regards, -- Benjamin Hahn <B.Hahn(a)phytec.de>

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[PATCH] ASoC: fsl: Use maple tree register cache
by Mark Brown 30 Sep '24

30 Sep '24

Several of the NXP drivers use regmaps with a rbtree register cache. Since the maple tree cache is uisng a generally more modern data structure which makes implementation choices more suitable for modern systems let's convert these drivers to it. This should have no practical impact. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..910b4ce7671c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -169,7 +169,7 @@ static const struct regmap_config fsl_aud2htx_regmap_config = { .readable_reg = fsl_aud2htx_readable_reg, .volatile_reg = fsl_aud2htx_volatile_reg, .writeable_reg = fsl_aud2htx_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static const struct of_device_id fsl_aud2htx_dt_ids[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..5149e732f763 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1748,7 +1748,7 @@ static const struct regmap_config fsl_easrc_regmap_config = { .rd_table = &fsl_easrc_readable_table, .wr_table = &fsl_easrc_writeable_table, .volatile_table = &fsl_easrc_volatileable_table, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; #ifdef DEBUG diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..c0dc9bebb77b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -955,7 +955,7 @@ static const struct regmap_config fsl_micfil_regmap_config = { .readable_reg = fsl_micfil_readable_reg, .volatile_reg = fsl_micfil_volatile_reg, .writeable_reg = fsl_micfil_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; /* END OF REGMAP */ --- base-commit: 98f7e32f20d28ec452afb208f9cffc08448a2652 change-id: 20240924-asoc-imx-maple-abb62f013533 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH] ASoC: meson: axg-iface: set continuous rates
by Jerome Brunet 30 Sep '24

30 Sep '24

The axg TDM HW does not depend on a selected set of rates. The hardware itself, just takes an input clock and work with it, regardless of its rate. In this way, the rates TDM can take are continuous. What might force the use of specific rate are the PLL available as clock and/or the codecs facing the TDM HW. Either way, this constraint does not belong in the TDM interface driver. Allow any rate as far as TDM is concerned by setting SNDRV_PCM_RATE_CONTINUOUS with an interval it has been tested with. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 12 +++++++++--- sound/soc/meson/axg-tdm.h | 2 -- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 62057c71f742..09103eef2a97 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -442,14 +442,18 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_PAD, @@ -461,7 +465,9 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Loopback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_LOOPBACK, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 1a17f546ce6e..acfcd48f8a00 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -15,8 +15,6 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 -#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ --- base-commit: cd87a98b53518e44cf3c1a7c1c07c869ce33bf83 change-id: 20240920-asoc-axg-iface-continuous-b94cfa234787 Best regards, -- Jerome

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[PATCH] ASoC: tas2781: Fix redundant parameter assignment
by Tang Bin 30 Sep '24

30 Sep '24

In these functions, the variable 'rc' is redundant, thus remove it. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 35 ++++++++-------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a8d97dd7..68887799e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -650,7 +650,6 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; @@ -659,9 +658,7 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, @@ -673,7 +670,6 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG; @@ -681,9 +677,7 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, reg = TAS2563_RUNTIME_RE_REG; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, @@ -696,7 +690,6 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); @@ -707,9 +700,7 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, else return -1; dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, @@ -721,13 +712,10 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, @@ -739,13 +727,10 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_nop_get( @@ -1115,14 +1100,12 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); int dev_id = ucontrol->value.integer.value[0]; - int max = tas_priv->ndev - 1, rc; + int max = tas_priv->ndev - 1; dev_id = clamp(dev_id, 0, max); guard(mutex)(&tas_priv->codec_lock); - rc = tasdev_chn_switch(tas_priv, dev_id); - - return rc; + return tasdev_chn_switch(tas_priv, dev_id); } static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) @@ -1339,10 +1322,8 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) i++; } - rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + return snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls < i ? nctrls : i); - - return rc; } static void tasdevice_fw_ready(const struct firmware *fmw, -- 2.33.0

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[PATCH v2] ALSA: hda/conexant: fix some typos
by Oldherl Oh 30 Sep '24

30 Sep '24

Fix some typos in patch_conexant.c Signed-off-by: Oldherl Oh <me(a)oldherl.one> --- sound/pci/hda/patch_conexant.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e851785ff058..ade42a8209c2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -166,18 +166,18 @@ static void cxt_init_gpio_led(struct hda_codec *codec) static void cx_fixup_headset_recog(struct hda_codec *codec) { - unsigned int mic_persent; + unsigned int mic_present; /* fix some headset type recognize fail issue, such as EDIFIER headset */ - /* set micbiasd output current comparator threshold from 66% to 55%. */ + /* set micbias output current comparator threshold from 66% to 55%. */ snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); - /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor + /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register * value adjustment trim from 2.2K ohms to 2.0K ohms. */ snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10); /* fix reboot headset type recognize fail issue */ - mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); - if (mic_persent & AC_PINSENSE_PRESENCE) + mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (mic_present & AC_PINSENSE_PRESENCE) /* enable headset mic VREF */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); else @@ -249,9 +249,9 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled, - * the node 19 can only be config to microphone or disabled. - * Check hp&mic tag to process headset pulgin&plugout. + /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + * the node 19 can only be configured to microphone or disabled. + * Check hp&mic tag to process headset plugin & plugout. */ mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */ -- 2.46.2

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ucm2: USB-Audio: add Steinberg UR22C (USB0499:172f)
by GitHub pull_request - opened 30 Sep '24

30 Sep '24

alsa-project/alsa-ucm-conf pull request #446 was opened from CoelacanthusHex: Similar to UR24C and UR44C, but with 2 inputs 2 outputs. Signed-off-by: Celeste Liu <CoelacanthusHex(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 30 Sep '24

30 Sep '24

The patch "Fix noise from speakers on Lenovo IdeaPad 3 15IAU7" caused that the headset push button cannot work properly in case of ALC257. This patch reverted it to correct the side effect. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

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Fix some typos in hda/conexant
by Oldherl Oh 30 Sep '24

30 Sep '24

I'm trying to understand Conexant driver code in order to solve my microphone detection problem. As I am new to kernel development, I have made this simple typo-fixing-only patch as an exercise. Hope it is not disturbing.

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[PATCH v2] ALSA: Fix typos in comments across various files
by Yu Jiaoliang 30 Sep '24

30 Sep '24

This patch fixes typos in comments within the ALSA subsystem. These changes improve code readability without affecting functionality. Signed-off-by: Yu Jiaoliang <yujiaoliang(a)vivo.com> --- v2: Update commit message. --- sound/core/compress_offload.c | 2 +- sound/core/oss/rate.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/sound.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b8c0d6edbdd1..bdf1d78de833 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,7 +288,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; guard(mutex)(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been setup */ switch (stream->runtime->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 98269119347f..b56eeda5e30e 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -294,7 +294,7 @@ static int rate_action(struct snd_pcm_plugin *plugin, default: break; } - return 0; /* silenty ignore other actions */ + return 0; /* silently ignore other actions */ } int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 99e39b5359cc..b49bc89c0758 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3115,7 +3115,7 @@ struct snd_pcm_sync_ptr32 { } c; } __packed; -/* recalcuate the boundary within 32bit */ +/* recalculate the boundary within 32bit */ static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t boundary; diff --git a/sound/core/sound.c b/sound/core/sound.c index b9db9aa0bfcb..6531a67f13b3 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -133,7 +133,7 @@ static struct snd_minor *autoload_device(unsigned int minor) /* /dev/aloadSEQ */ snd_request_other(minor); } - mutex_lock(&sound_mutex); /* reacuire lock */ + mutex_lock(&sound_mutex); /* reacquire lock */ return snd_minors[minor]; } #else /* !CONFIG_MODULES */ -- 2.34.1

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[PATCH v1] slimbus: Correct some typos in comments
by Shen Lichuan 29 Sep '24

29 Sep '24

Fixed some confusing typos that were currently identified with codespell, the details are as follows: -in the code comments: drivers/slimbus/messaging.c:226: transfered ==> transferred drivers/slimbus/messaging.c:230: vlaue ==> value drivers/slimbus/qcom-ctrl.c:127: Oder ==> Order drivers/slimbus/slimbus.h:263: Presense ==> Presence drivers/slimbus/slimbus.h:371: resposible ==> responsible Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/slimbus/messaging.c | 4 ++-- drivers/slimbus/qcom-ctrl.c | 2 +- drivers/slimbus/slimbus.h | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index 242570a5e565..b7dd2badc9d1 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -223,11 +223,11 @@ static u16 slim_slicesize(int code) /** * slim_xfer_msg() - Transfer a value info message on slim device * - * @sbdev: slim device to which this msg has to be transfered + * @sbdev: slim device to which this msg has to be transferred * @msg: value info message pointer * @mc: message code of the message * - * Called by drivers which want to transfer a vlaue or info elements. + * Called by drivers which want to transfer a value or info elements. * * Return: -ETIMEDOUT: If transmission of this message timed out */ diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index e25f9bdd9b23..039384d85e7b 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -124,7 +124,7 @@ static void qcom_slim_queue_tx(struct qcom_slim_ctrl *ctrl, void *buf, __iowrite32_copy(ctrl->base + tx_reg, buf, count); - /* Ensure Oder of subsequent writes */ + /* Ensure Order of subsequent writes */ mb(); } diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..2067d47541f2 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -260,7 +260,7 @@ enum slim_ch_aux_bit_fmt { * struct slim_channel - SLIMbus channel, used for state machine * * @id: ID of channel - * @prrate: Presense rate of channel from Table 66 of SLIMbus 2.0 Specs + * @prrate: Presence rate of channel from Table 66 of SLIMbus 2.0 Specs * @seg_dist: segment distribution code from Table 20 of SLIMbus 2.0 Specs * @data_fmt: Data format of channel. * @aux_fmt: Aux format for this channel. @@ -368,7 +368,7 @@ struct slim_stream_runtime { * @get_laddr: It is possible that controller needs to set fixed logical * address table and get_laddr can be used in that case so that controller * can do this assignment. Use case is when the master is on the remote - * processor side, who is resposible for allocating laddr. + * processor side, who is responsible for allocating laddr. * @wakeup: This function pointer implements controller-specific procedure * to wake it up from clock-pause. Framework will call this to bring * the controller out of clock pause. -- 2.17.1

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