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

[PATCH v1] ASoC: codecs: lpass-macro: Add support for channel map mixer control
by Mohammad Rafi Shaik 08 Oct '24

08 Oct '24

From: Rohit kumar <quic_rohkumar(a)quicinc.com> Add channel map mixer control for lpass macro codec DAIs. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> Co-developed-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> Signed-off-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> --- sound/soc/codecs/lpass-macro-common.c | 55 +++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 2 + sound/soc/codecs/lpass-rx-macro.c | 12 ++++++ sound/soc/codecs/lpass-tx-macro.c | 9 +++++ sound/soc/codecs/lpass-va-macro.c | 9 +++++ sound/soc/codecs/lpass-wsa-macro.c | 12 ++++++ 6 files changed, 99 insertions(+) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 6e3b8d0897dd..777af4885245 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -8,12 +8,67 @@ #include <linux/platform_device.h> #include <linux/pm_domain.h> #include <linux/pm_runtime.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> #include "lpass-macro-common.h" static DEFINE_MUTEX(lpass_codec_mutex); static enum lpass_codec_version lpass_codec_version; +static int lpass_macro_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_soc_dai *dai = info->private_data; + u32 *chmap_data = NULL; + u32 rx_ch_cnt = 0; + u32 tx_ch_cnt = 0; + u32 rx_ch, tx_ch; + + chmap_data = kzalloc(sizeof(u32) * 2, GFP_KERNEL); + if (!chmap_data) + return -ENOMEM; + + snd_soc_dai_get_channel_map(dai, &tx_ch_cnt, &tx_ch, &rx_ch_cnt, &rx_ch); + if (rx_ch_cnt) { + chmap_data[0] = rx_ch_cnt; + chmap_data[1] = rx_ch; + } else if (tx_ch_cnt) { + chmap_data[0] = tx_ch_cnt; + chmap_data[1] = tx_ch; + } + memcpy(ucontrol->value.bytes.data, chmap_data, sizeof(u32) * 2); + + kfree(chmap_data); + return 0; +} + +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir) +{ + struct snd_pcm_chmap *info; + int ret; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, dir, NULL, + 2 * sizeof(u32), 0, &info); + if (ret < 0) { + kfree(info); + return ret; + } + + /* override handlers */ + info->private_data = dai; + info->kctl->get = lpass_macro_chmap_ctl_get; + return 0; +} +EXPORT_SYMBOL_GPL(lpass_macro_add_chmap_ctls); + struct lpass_macro *lpass_macro_pds_init(struct device *dev) { struct lpass_macro *l_pds; diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index fb4b96cb2b23..23ed6836addf 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -40,6 +40,8 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev); void lpass_macro_pds_exit(struct lpass_macro *pds); void lpass_macro_set_codec_version(enum lpass_codec_version version); enum lpass_codec_version lpass_macro_get_codec_version(void); +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir); static inline void lpass_macro_pds_exit_action(void *pds) { diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 71e0d3bffd3f..05bd0f8d126b 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -1954,10 +1954,22 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int rx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == RX_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops rx_macro_dai_ops = { .hw_params = rx_macro_hw_params, .get_channel_map = rx_macro_get_channel_map, .mute_stream = rx_macro_digital_mute, + .pcm_new = rx_macro_pcm_new, }; static struct snd_soc_dai_driver rx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index a134584acf90..c7a8e2694e36 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1211,10 +1211,19 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int tx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops tx_macro_dai_ops = { .hw_params = tx_macro_hw_params, .get_channel_map = tx_macro_get_channel_map, .mute_stream = tx_macro_digital_mute, + .pcm_new = tx_macro_pcm_new, }; static struct snd_soc_dai_driver tx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index c781da476240..1a13f472ed3a 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -939,10 +939,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int va_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops va_macro_dai_ops = { .hw_params = va_macro_hw_params, .get_channel_map = va_macro_get_channel_map, .mute_stream = va_macro_digital_mute, + .pcm_new = va_macro_pcm_new, }; static struct snd_soc_dai_driver va_macro_dais[] = { diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index c989d82d1d3c..7c4fd4bf2668 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1344,9 +1344,21 @@ static int wsa_macro_get_channel_map(const struct snd_soc_dai *dai, return 0; } +static int wsa_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == WSA_MACRO_AIF_VI || dai->id == WSA_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops wsa_macro_dai_ops = { .hw_params = wsa_macro_hw_params, .get_channel_map = wsa_macro_get_channel_map, + .pcm_new = wsa_macro_pcm_new, }; static struct snd_soc_dai_driver wsa_macro_dai[] = { -- 2.25.1

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[PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 05 Oct '24

05 Oct '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

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[PATCH v2] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 04 Oct '24

04 Oct '24

The headset push button cannot work properly in case of the ALC257. This patch reverted the previous commit to correct the side effect. Fixes: ef9718b3d54e ("ALSA: hda/realtek: Fix noise from speakers on Lenovo IdeaPad 3 15IAU7") 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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[PATCH v1] soundwire: Correct some typos in comments
by Shen Lichuan 03 Oct '24

03 Oct '24

Fixed some confusing spelling errors that were currently identified, the details are as follows: -in the code comments: drivers/soundwire/bus.c:115: enumuration==> enumeration drivers/soundwire/qcom.c:1176: wil ==> will drivers/soundwire/sysfs_slave.c:218: highligh ==> highlight Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/soundwire/bus.c | 2 +- drivers/soundwire/qcom.c | 2 +- drivers/soundwire/sysfs_slave.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 263ca32f0c5c..d1dc62c34f1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -112,7 +112,7 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* Set higher order bits */ *bus->assigned = ~GENMASK(SDW_BROADCAST_DEV_NUM, SDW_ENUM_DEV_NUM); - /* Set enumuration device number and broadcast device number */ + /* Set enumeration device number and broadcast device number */ set_bit(SDW_ENUM_DEV_NUM, bus->assigned); set_bit(SDW_BROADCAST_DEV_NUM, bus->assigned); diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2b403b14066c 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1173,7 +1173,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, else sconfig.direction = SDW_DATA_DIR_RX; - /* hw parameters wil be ignored as we only support PDM */ + /* hw parameters will be ignored as we only support PDM */ sconfig.ch_count = 1; sconfig.frame_rate = params_rate(params); sconfig.type = stream->type; diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index f4259710dd0f..c5c22d1708ec 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -215,7 +215,7 @@ const struct attribute_group *sdw_attr_groups[] = { /* * the status is shown in capital letters for UNATTACHED and RESERVED - * on purpose, to highligh users to the fact that these status values + * on purpose, to highlight users to the fact that these status values * are not expected. */ static const char *const slave_status[] = { -- 2.17.1

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[PATCH 0/4] refactor amd soundwire manager driver for acp 6.3 platform
by Vijendar Mukunda 03 Oct '24

03 Oct '24

This patch series refactors the amd soundwire manager driver to support acp 6.3 platform based on acp pci revision id. This patch series prepared on top of asoc tree for-next branch. This patch series should be applied at one go. @Vinod: Please help to review soundwire patches and provide ack-by for this patch series. Vijendar Mukunda (4): soundwire: amd: pass acp pci revision id as resource data soundwire: amd: refactor existing code for acp 6.3 platform ASoC: SOF: amd: pass acp_rev as soundwire resource data ASoC: amd: ps: pass acp pci revision id as soundwire resource data drivers/soundwire/amd_init.c | 1 + drivers/soundwire/amd_manager.c | 99 ++++++++++++++++++++----------- drivers/soundwire/amd_manager.h | 16 ++--- include/linux/soundwire/sdw_amd.h | 6 ++ sound/soc/amd/ps/acp63.h | 2 + sound/soc/amd/ps/pci-ps.c | 2 + sound/soc/sof/amd/acp.c | 1 + 7 files changed, 84 insertions(+), 43 deletions(-) -- 2.34.1

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[PATCH] ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value
by Charles Han 02 Oct '24

02 Oct '24

devm_kasprintf() can return a NULL pointer on failure but this returned value is not checked. Fixes: b359760d95ee ("ASoC: intel: sof_sdw: Add simple DAI link creation helper") Signed-off-by: Charles Han <hanchunchao(a)inspur.com> --- sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf..26917f6f15cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -919,6 +919,9 @@ static int create_ssp_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", ssp_info->acpi_id, j++); + if (!name || !cpu_dai_name || !codec_name) + return -ENOMEM; + int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; @@ -985,6 +988,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, for (i = 0; i < hdmi_num; i++) { char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); + if (!name || !cpu_dai_name) + return -ENOMEM; + char *codec_name, *codec_dai_name; if (intel_ctx->hdmi.idisp_codec) { @@ -996,6 +1002,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, codec_dai_name = "snd-soc-dummy-dai"; } + if (!codec_dai_name) + return -ENOMEM; + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, @@ -1019,6 +1028,9 @@ static int create_bt_dailinks(struct snd_soc_card *card, SOF_BT_OFFLOAD_SSP_SHIFT; char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); + if (!name || !cpu_dai_name) + return -ENOMEM; + int ret; ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, -- 2.31.1

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[RFC PATCH v3 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 02 Oct '24

02 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH] Fix possible NULL Pointer Dereference in 'asoc_qcom_lpass_cpu_platform_probe'
by Gax-c 01 Oct '24

01 Oct '24

A 'devm_kzalloc' in 'asoc_qcom_lpass_cpu_platform_probe' could possibly return NULL pointer. NULL Pointer Dereference may be triggerred in 'asoc_qcom_lpass_cpu_platform_probe' without addtional check. Add a null check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zijie Zhao <zzjas98(a)gmail.com> Reported-by: Chenyuan Yang <chenyuan0y(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..a8e56f47f237 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1243,6 +1243,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; + /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); -- 2.25.1

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[PATCH 0/3] ASoC: fsl_micfil: fix and improvement
by Shengjiu Wang 01 Oct '24

01 Oct '24

Fix the usage of regmap_write_bits(). Move mclk clock enablement to late stage. Enable the micfil error interrupt. Shengjiu Wang (3): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt sound/soc/fsl/fsl_micfil.c | 49 ++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

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