December 2023 - Alsa-devel - mailman.alsa-project.org

[RFC, PATCH 1/2] ASoC: qcom: sdw: Add TDM support
by Jianhua Lu 11 Dec '23

11 Dec '23

Add qcom TDM setup function to support TDM ports for qcom platform. Signed-off-by: Jianhua Lu <lujianhua000(a)gmail.com> --- sound/soc/qcom/sdw.c | 64 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c index 77dbe0c28b29..267a8cff01a4 100644 --- a/sound/soc/qcom/sdw.c +++ b/sound/soc/qcom/sdw.c @@ -4,6 +4,7 @@ #include <dt-bindings/sound/qcom,q6afe.h> #include <linux/module.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include "sdw.h" @@ -101,6 +102,66 @@ int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare); +static int qcom_snd_tdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + unsigned int tdm_slot_offset[8] = { 0, 4, 8, 12, 16, 20, 24, 28 }; + int channels, slot_width; + int ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + slot_width = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + slot_width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + slot_width = 32; + break; + default: + dev_err(rtd->dev, "%s: invalid param format 0x%x\n", __func__, + params_format(params)); + return -EINVAL; + } + + channels = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, channels, + tdm_slot_offset); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + return ret; + } + } else { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xf, 0, 8, slot_width); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set tdm slot, err:%d\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_channel_map(cpu_dai, channels, + tdm_slot_offset, 0, NULL); + if (ret < 0) { + dev_err(rtd->dev, "%s: failed to set channel map, err:%d\n", + __func__, ret); + return ret; + } + } +} +EXPORT_SYMBOL_GPL(qcom_snd_tdm_hw_params); + int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct sdw_stream_runtime **psruntime) @@ -125,6 +186,9 @@ int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, *psruntime = sruntime; } break; + case PRIMARY_TDM_RX_0...QUINARY_TDM_TX_7: + qcom_snd_tdm_hw_params(substream, params); + break; } return 0; -- 2.41.0

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[PATCH] ASoC: codec: wsa884x: make use of new mute_unmute_on_trigger flag
by Neil Armstrong 11 Dec '23

11 Dec '23

This fix is based on commit [1] fixing click and pop sounds during SoundWire port start because PA is left unmuted. making use of new mute_unmute_on_trigger flag and removing unmute at PA setup, removes the Click/Pop issue at SoundWire enable. [1] 805ce81826c8 ("ASoC: codecs: wsa883x: make use of new mute_unmute_on_trigger flag") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: aa21a7d4f68a ("ASoC: codecs: wsa884x: Add WSA884x family of speakers") Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- sound/soc/codecs/wsa884x.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 993d76b18b53..f2653df84e4a 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -1654,15 +1654,9 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x1); - snd_soc_component_write_field(component, WSA884X_PA_FSM_EN, - WSA884X_PA_FSM_EN_GLOBAL_PA_EN_MASK, - 0x1); break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_component_write_field(component, WSA884X_PA_FSM_EN, - WSA884X_PA_FSM_EN_GLOBAL_PA_EN_MASK, - 0x0); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); @@ -1786,6 +1780,7 @@ static const struct snd_soc_dai_ops wsa884x_dai_ops = { .hw_free = wsa884x_hw_free, .mute_stream = wsa884x_mute_stream, .set_stream = wsa884x_set_stream, + .mute_unmute_on_trigger = true, }; static struct snd_soc_dai_driver wsa884x_dais[] = { --- base-commit: bbd220ce4e29ed55ab079007cff0b550895258eb change-id: 20231211-topic-sm8x50-upstream-wsa884x-fix-plop-a7a7ca7ec21c Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

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[PATCH v2 1/3] ASoC: qcom: audioreach: Commonize setting channel mappings
by Krzysztof Kozlowski 11 Dec '23

11 Dec '23

Move code assigning channel mapping values to a common helper function. This simplifies three out of four cases, with the last case using incompatible type (uint16_t array instead of uint8_t array). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. New patch --- sound/soc/qcom/qdsp6/audioreach.c | 35 ++++++++++++++----------------- 1 file changed, 16 insertions(+), 19 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5974c7929dd3..3db5ff367a29 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -267,6 +267,16 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token } EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt); +static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) +{ + if (num_channels == 1) { + ch_map[0] = PCM_CHANNEL_L; + } else if (num_channels == 2) { + ch_map[0] = PCM_CHANNEL_L; + ch_map[1] = PCM_CHANNEL_R; + } +} + static void apm_populate_container_config(struct apm_container_obj *cfg, struct audioreach_container *cont) { @@ -864,12 +874,8 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr, mp3_cfg->endianness = PCM_LITTLE_ENDIAN; mp3_cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (mcfg->num_channels == 2) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(mp3_cfg->channel_mapping, + mcfg->num_channels); break; case SND_AUDIOCODEC_AAC: media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED; @@ -1089,13 +1095,8 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, media_cfg->q_factor = mcfg->bit_width - 1; media_cfg->bits_per_sample = mcfg->bit_width; - if (num_channels == 1) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (num_channels == 2) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - media_cfg->channel_mapping[1] = PCM_CHANNEL_R; - - } + audioreach_set_channel_mapping(media_cfg->channel_mapping, + num_channels); rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -1153,12 +1154,8 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, cfg->endianness = PCM_LITTLE_ENDIAN; cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) - cfg->channel_mapping[0] = PCM_CHANNEL_L; - else if (num_channels == 2) { - cfg->channel_mapping[0] = PCM_CHANNEL_L; - cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(cfg->channel_mapping, + num_channels); } else { rc = audioreach_set_compr_media_format(header, p, mcfg); if (rc) { -- 2.34.1

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[PATCH 1/4] ASoC: dt-bindings: qcom,lpass-rx-macro: Add SM8650 LPASS RX
by Krzysztof Kozlowski 11 Dec '23

11 Dec '23

Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) RX macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong <neil.armstrong(a)linaro.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../bindings/sound/qcom,lpass-rx-macro.yaml | 21 ++++++++++++------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index ec4b0ac8ad68..cbc36646100f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -11,12 +11,16 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-rx-macro - - qcom,sm8250-lpass-rx-macro - - qcom,sm8450-lpass-rx-macro - - qcom,sm8550-lpass-rx-macro - - qcom,sc8280xp-lpass-rx-macro + oneOf: + - enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sm8550-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + - items: + - const: qcom,sm8650-lpass-rx-macro + - const: qcom,sm8550-lpass-rx-macro reg: maxItems: 1 @@ -96,8 +100,9 @@ allOf: - if: properties: compatible: - enum: - - qcom,sm8550-lpass-rx-macro + contains: + enum: + - qcom,sm8550-lpass-rx-macro then: properties: clocks: -- 2.34.1

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[PATCH AUTOSEL 4.14 1/5] ASoC: cs43130: Fix the position of const qualifier
by Sasha Levin 11 Dec '23

11 Dec '23

From: Maciej Strozek <mstrozek(a)opensource.cirrus.com> [ Upstream commit e7f289a59e76a5890a57bc27b198f69f175f75d9 ] Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20231117141344.64320-2-mstrozek@opensource.cirrus… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 643e37fc218e1..793496e8d262d 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1675,7 +1675,7 @@ static ssize_t cs43130_show_dc_r(struct device *dev, return cs43130_show_dc(dev, buf, HP_RIGHT); } -static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { +static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = { 24, 43, 93, @@ -2351,7 +2351,7 @@ static const struct regmap_config cs43130_regmap = { .use_single_rw = true, /* needed for regcache_sync */ }; -static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { +static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 50, 120, }; -- 2.42.0

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[PATCH AUTOSEL 4.19 1/7] ASoC: Intel: Skylake: mem leak in skl register function
by Sasha Levin 11 Dec '23

11 Dec '23

From: Kamil Duljas <kamil.duljas(a)gmail.com> [ Upstream commit f8ba14b780273fd290ddf7ee0d7d7decb44cc365 ] skl_platform_register() uses krealloc. When krealloc is fail, then previous memory is not freed. The leak is also when soc component registration failed. Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20231116224112.2209-2-kamil.duljas@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 6b2c8c6e7a00f..5195e012dc6d4 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1450,6 +1450,7 @@ int skl_platform_register(struct device *dev) dais = krealloc(skl->dais, sizeof(skl_fe_dai) + sizeof(skl_platform_dai), GFP_KERNEL); if (!dais) { + kfree(skl->dais); ret = -ENOMEM; goto err; } @@ -1462,8 +1463,10 @@ int skl_platform_register(struct device *dev) ret = devm_snd_soc_register_component(dev, &skl_component, skl->dais, num_dais); - if (ret) + if (ret) { + kfree(skl->dais); dev_err(dev, "soc component registration failed %d\n", ret); + } err: return ret; } -- 2.42.0

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[PATCH AUTOSEL 5.4 01/12] ASoC: Intel: Skylake: Fix mem leak in few functions
by Sasha Levin 11 Dec '23

11 Dec '23

From: Kamil Duljas <kamil.duljas(a)gmail.com> [ Upstream commit d5c65be34df73fa01ed05611aafb73b440d89e29 ] The resources should be freed when function return error. Signed-off-by: Kamil Duljas <kamil.duljas(a)gmail.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20231116125150.1436-1-kamil.duljas@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 4 +++- sound/soc/intel/skylake/skl-sst-ipc.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 439dd4ba690c4..19176ae272742 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -260,8 +260,10 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (!mconfig) + if (!mconfig) { + kfree(dma_params); return -EINVAL; + } skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 667cdddc289fd..7286cbd0c46f9 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -1003,8 +1003,10 @@ int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; buf = krealloc(reply.data, reply.size, GFP_KERNEL); - if (!buf) + if (!buf) { + kfree(reply.data); return -ENOMEM; + } *payload = buf; *bytes = reply.size; -- 2.42.0

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[PATCH AUTOSEL 5.10 08/16] ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available
by Sasha Levin 11 Dec '23

11 Dec '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit 3d1dc8b1030df8ca0fdfd4905c88ee10db943bf8 ] When the HDMI is not present due to disabled display support we will use dummy codec and the HDMI routes will refer to non existent DAPM widgets. Trim the route list from the HDMI routes to be able to probe the card even if the HDMI dais are not registered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20231124124015.15878-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index bc50eda297ab7..9d0f08db709a5 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -157,6 +157,8 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); if (!ctx->idisp_codec) { + card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; + num_route -= IDISP_ROUTE_COUNT; for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = dummy_codec; skl_hda_be_dai_links[i].num_codecs = -- 2.42.0

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[PATCH AUTOSEL 5.10 06/16] ASoC: cs43130: Fix incorrect frame delay configuration
by Sasha Levin 11 Dec '23

11 Dec '23

From: Maciej Strozek <mstrozek(a)opensource.cirrus.com> [ Upstream commit aa7e8e5e4011571022dc06e4d7a2f108feb53d1a ] Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20231117141344.64320-3-mstrozek@opensource.cirrus… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 285806868c405..02fb9317b6970 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -578,7 +578,7 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, break; case SND_SOC_DAIFMT_LEFT_J: hi_size = bitwidth_sclk; - frm_delay = 2; + frm_delay = 0; frm_phase = 1; break; case SND_SOC_DAIFMT_DSP_A: -- 2.42.0

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[PATCH AUTOSEL 5.10 05/16] ASoC: cs43130: Fix the position of const qualifier
by Sasha Levin 11 Dec '23

11 Dec '23

From: Maciej Strozek <mstrozek(a)opensource.cirrus.com> [ Upstream commit e7f289a59e76a5890a57bc27b198f69f175f75d9 ] Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20231117141344.64320-2-mstrozek@opensource.cirrus… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 8f70dee958786..285806868c405 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1683,7 +1683,7 @@ static ssize_t cs43130_show_dc_r(struct device *dev, return cs43130_show_dc(dev, buf, HP_RIGHT); } -static u16 const cs43130_ac_freq[CS43130_AC_FREQ] = { +static const u16 cs43130_ac_freq[CS43130_AC_FREQ] = { 24, 43, 93, @@ -2364,7 +2364,7 @@ static const struct regmap_config cs43130_regmap = { .use_single_write = true, }; -static u16 const cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { +static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 50, 120, }; -- 2.42.0

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