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

[PATCH AUTOSEL 5.10 03/14] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 1d049685e7075..47b581d99da67 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -468,6 +468,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

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[PATCH AUTOSEL 5.15 06/17] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 3328c02f93c74..1dfe439d13417 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -353,11 +353,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

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[PATCH AUTOSEL 5.15 05/17] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 5b82329f44401..dbd30604816e5 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

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[PATCH AUTOSEL 5.15 03/17] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 434679afa7e1a..3d2a0e8cad9a5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -607,6 +607,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

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[PATCH AUTOSEL 6.1 09/27] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 0dc0475670ffe..554e7896e8053 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

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[PATCH AUTOSEL 6.1 08/27] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 4edf5b27e136b..c6ef8f92b25f1 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

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[PATCH AUTOSEL 6.1 06/27] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d6ef8e850412b..ff879e173d51d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

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[PATCH AUTOSEL 6.6 11/33] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index a3663ab065ac2..0a731b21e5a58 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

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[PATCH AUTOSEL 6.6 10/33] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a5c2cca38d01a..8c8b2a2f6f862 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1474,10 +1474,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1492,6 +1493,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1591,9 +1593,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

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[PATCH AUTOSEL 6.6 08/33] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 651408c6f399d..5b8b21ade9cfe 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

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[PATCH AUTOSEL 6.9 18/40] ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
by Sasha Levin 09 Jul '24

09 Jul '24

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 8af49868e51ed1ba117b74728af12abe1eda82e5 ] If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer sources should be initialized to disconnected. There aren't currently any available products using the ASP so this doesn't affect any existing systems. The cs35l56 does not have any fixed default for the mixer source registers. When the cs35l56 boots, its firmware patches these registers to setup a system-specific routing; this is so that Windows can use generic SDCA drivers instead of needing knowledge of chip-specific registers. The setup varies between end-products, which each have customized firmware, and so the default register state varies between end-products. It can also change if the firmware on an end-product is upgraded - for example if a change was needed to the routing for Windows use-cases. It must be emphasized that the settings applied by the firmware are not internal magic tuning; they are statically implementing use-case setup that on Linux would be done via ALSA controls. The driver is currently syncing the mixer controls with whatever initial state the firmware wrote to the registers, so that they report the actual audio routing. But if the ASP DAI is hooked up this can create a powered-up DAPM graph without anything intentionally setting up a path. This can lead to parts of the audio system powering up unexpectedly. For example when cs35l56 is connected to cs42l43 using a codec-codec link, this can create a complete DAPM graph which then powers-up cs42l43. But the cs42l43 can only be clocked from its SoundWire bus so this causes a bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up without a clock. If the host is taking ownership of the ASP (either directly or as a codec-to-codec link) there is no need to keep the mixer settings that the firmware wrote. The driver has ALSA controls for setting these using standard Linux mechanisms. So if the machine driver hooks up the ASP the ASP mixers are initialized to "None" (no input). This prevents unintended DAPM-graph power-ups, and means the initial state of the mixers is always going to be None. Since the initial state of the mixers can vary from system to system and potentially between firmware upgrades, no use-case manager can currently assume that cs35l56 has a known initial state. The firmware could just as easily default them to "None" as to any input source. So defaulting them to "None" in the driver is not increasing the entropy of the system. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l56-shared.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index fd02b621da52c..d29878af2a80d 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -214,6 +214,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = { REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), + REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000), }; /* -- 2.43.0

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[PATCH AUTOSEL 6.9 14/40] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 4513b527ab970..ad8925b6481ca 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 13/40] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 1e760c3155213..2b1ed91a736c9 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 11/40] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b41a1147f1c34..a64d1989e28a5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 08/40] ASoC: Intel: avs: Fix route override
by Sasha Levin 09 Jul '24

09 Jul '24

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit fd660b1bd015e5aa9a558ee04088f2431010548d ] Instead of overriding existing memory strings that may be too short, just allocate needed memory and point the route at it. Reported-by: Jason Montleon <jmontleo(a)redhat.com> Link: https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-21… Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20240603102818.36165-3-amadeuszx.slawinski@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/topology.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 42b42903ae9de..691d16ce95a0f 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1545,8 +1545,8 @@ static int avs_route_load(struct snd_soc_component *comp, int index, { struct snd_soc_acpi_mach *mach = dev_get_platdata(comp->card->dev); size_t len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN; - char buf[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ssp_port, tdm_slot; + char *buf; /* See parse_link_formatted_string() for dynamic naming when(s). */ if (!avs_mach_singular_ssp(mach)) @@ -1557,13 +1557,24 @@ static int avs_route_load(struct snd_soc_component *comp, int index, return 0; tdm_slot = avs_mach_ssp_tdm(mach, ssp_port); + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->source, ssp_port, tdm_slot); - strscpy((char *)route->source, buf, len); + route->source = buf; + + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->sink, ssp_port, tdm_slot); - strscpy((char *)route->sink, buf, len); + route->sink = buf; + if (route->control) { + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->control, ssp_port, tdm_slot); - strscpy((char *)route->control, buf, len); + route->control = buf; } return 0; -- 2.43.0

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my headphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Enable Mute LED on HP 250 G7
by Nazar Bilinskyi 09 Jul '24

09 Jul '24

HP 250 G7 has a mute LED that can be made to work using quirk ALC269_FIXUP_HP_LINE1_MIC1_LED. Enable already existing quirk. Signed-off-by: Nazar Bilinskyi <nbilinskyi(a)gmail.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 811e82474..57eb67f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10053,6 +10053,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84a6, "HP 250 G7 Notebook PC", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.45.2

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - opened 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was opened from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: (subset) [PATCH v6 0/7] ASoC: fsl-asoc-card: add S/PDIF controller support
by Mark Brown 08 Jul '24

08 Jul '24

On Thu, 27 Jun 2024 10:30:57 +0200, Elinor Montmasson wrote: > This is the v6 of the series of patches aiming to make the machine > driver `fsl-asoc-card` compatible with S/PDIF controllers on imx boards. > The main goal is to allow the use of S/PDIF controllers with ASRC > modules. > > The `imx-spdif` machine driver already has specific support for S/PDIF > controllers but doesn't support using an ASRC with it. However, the > `fsl-asoc-card` machine driver has the necessary code to create a sound > card which can use an ASRC module. > It is then possible to extend the support for S/PDIF audio cards by > merging the `imx-spdif` driver into `fsl-asoc-card`. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/7] ASoC: fsl-asoc-card: add support for dai links with multiple codecs commit: a613b63db233b6e7d46ec72f009c1cbb7db8be66 [2/7] ASoC: fsl-asoc-card: add second dai link component for codecs commit: c68fa0d9b0f8cc7c3ae7d29c02adbc97622a73f5 [3/7] ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links commit: fcc6ace84f1f6ce2211af25c3c8fb30a0fb2bb2c [4/7] ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c commit: 6d174cc4f22461ad3fe383570527e86bf1948a2e [5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge commit: 4359caadd17fbde64d656c10bd6f2dc91b675a11 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH alsa-ucm-conf v3 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

2 3

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No change since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

2 3

[PATCH] firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files
by Richard Fitzgerald 08 Jul '24

08 Jul '24

Use strnlen() instead of strlen() on the algorithm and coefficient name string arrays in V1 wmfw files. In V1 wmfw files the name is a NUL-terminated string in a fixed-size array. cs_dsp should protect against overrunning the array if the NUL terminator is missing. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs") --- drivers/firmware/cirrus/cs_dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index d2aa0980ed78..0a504a3c4375 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1166,7 +1166,7 @@ static inline void cs_dsp_coeff_parse_alg(struct cs_dsp *dsp, const u8 **data, blk->id = le32_to_cpu(raw->id); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ncoeff = le32_to_cpu(raw->ncoeff); break; default: @@ -1199,7 +1199,7 @@ static inline void cs_dsp_coeff_parse_coeff(struct cs_dsp *dsp, const u8 **data, blk->offset = le16_to_cpu(raw->hdr.offset); blk->mem_type = le16_to_cpu(raw->hdr.type); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ctl_type = le16_to_cpu(raw->ctl_type); blk->flags = le16_to_cpu(raw->flags); blk->len = le32_to_cpu(raw->len); -- 2.39.2

2 5

[PATCH v4 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 08 Jul '24

08 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Olivier Moysan (8): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 23 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 153 ++++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 298 +++++++++++++++--- drivers/iio/industrialio-backend.c | 116 ++++++- include/linux/iio/backend.h | 14 +- 7 files changed, 633 insertions(+), 65 deletions(-) base-commit: 529d2e1900642eba6df28307e26e19793e227546 -- 2.25.1

2 2

[PATCH 00/11] ASoC: Simplify code with cleanup.h
by Krzysztof Kozlowski 08 Jul '24

08 Jul '24

Allocate the memory with scoped/cleanup.h to reduce error handling (simpler error paths) and make the code a bit smaller. Best regards, Krzysztof --- Krzysztof Kozlowski (11): ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_add_dapms() with cleanup.h ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_probe() with cleanup.h ASoC: codecs: wcd9335: Simplify with cleanup.h ASoC: codecs: wcd934x: Simplify with cleanup.h ASoC: simple-card-utils: Simplify with cleanup.h ASoC: audio-graph-card: Use cleanup.h instead of devm_kfree() ASoC: audio-graph-card2: Use cleanup.h instead of devm_kfree() ASoC: simple-card: Use cleanup.h instead of devm_kfree() ASoC: ops: Simplify with cleanup.h ASoC: dapm: Simplify dapm_cnew_widget() with cleanup.h ASoC: dapm: Simplify snd_soc_dai_link_event_pre_pmu() with cleanup.h sound/soc/codecs/audio-iio-aux.c | 83 ++++++++++++----------------------- sound/soc/codecs/wcd9335.c | 22 ++++------ sound/soc/codecs/wcd934x.c | 22 ++++------ sound/soc/generic/audio-graph-card.c | 5 +-- sound/soc/generic/audio-graph-card2.c | 5 +-- sound/soc/generic/simple-card-utils.c | 13 +++--- sound/soc/generic/simple-card.c | 5 +-- sound/soc/soc-dapm.c | 47 ++++++++------------ sound/soc/soc-ops.c | 26 +++++------ 9 files changed, 87 insertions(+), 141 deletions(-) --- base-commit: cf601827c51b0496f6bb8a1865f1bf69ebd246b7 change-id: 20240703-asoc-cleanup-h-ef9cbfb67b66 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 16

[PATCH v2] firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files
by Richard Fitzgerald 08 Jul '24

08 Jul '24

Use strnlen() instead of strlen() on the algorithm and coefficient name string arrays in V1 wmfw files. In V1 wmfw files the name is a NUL-terminated string in a fixed-size array. cs_dsp should protect against overrunning the array if the NUL terminator is missing. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs") --- drivers/firmware/cirrus/cs_dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 16484ab9b09d..8a347b938406 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1183,4 +1183,4 @@ static int cs_dsp_coeff_parse_alg(struct cs_dsp *dsp, blk->id = le32_to_cpu(raw->id); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ncoeff = le32_to_cpu(raw->ncoeff); @@ -1260,4 +1260,4 @@ static int cs_dsp_coeff_parse_coeff(struct cs_dsp *dsp, blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ctl_type = le16_to_cpu(raw->ctl_type); blk->flags = le16_to_cpu(raw->flags); -- 2.39.2

1 1

[PATCH v6 5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge
by Elinor Montmasson 08 Jul '24

08 Jul '24

The S/PDIF audio card support with compatible "fsl,imx-audio-spdif" was merged from imx-spdif into the fsl-asoc-card driver. It makes possible to use an S/PDIF with an ASRC. This merge introduces new DT bindings to use with compatible "fsl,imx-audio-spdif" to follow the way fsl-asoc-card works: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". In an upcoming commit, in-tree DTs will be modified to follow these new properties: * Property "spdif-controller" will be renamed "audio-cpu". * spdif_transmitter and spdif_receiver nodes will be declared and linked to the fsl-asoc-card node with the property "audio-codec". To keep backward compatibility with other DTs, support for "spdif-controller", "spdif-in" and "spdif-out" properties is kept. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still supported, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ------------------- .../bindings/sound/fsl-asoc-card.yaml | 53 +++++++++++++-- 2 files changed, 49 insertions(+), 70 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Freescale i.MX audio complex with S/PDIF transceiver - -maintainers: - - Shengjiu Wang <shengjiu.wang(a)nxp.com> - -properties: - compatible: - oneOf: - - items: - - enum: - - fsl,imx-sabreauto-spdif - - fsl,imx6sx-sdb-spdif - - const: fsl,imx-audio-spdif - - enum: - - fsl,imx-audio-spdif - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - spdif-controller: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of the i.MX S/PDIF controller - - spdif-out: - type: boolean - description: - If present, the transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector or jack on the - board or it's connecting to some other IP block, such as an HDMI - encoder or display-controller. - - spdif-in: - type: boolean - description: - If present, the receiving function of S/PDIF will be enabled, - indicating there is a physical S/PDIF in connector/jack on the board. - -required: - - compatible - - model - - spdif-controller - -anyOf: - - required: - - spdif-in - - required: - - spdif-out - -additionalProperties: false - -examples: - - | - sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..92aa47ec72c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -65,6 +65,11 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +86,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +99,15 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +163,10 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. + minItems: 1 + maxItems: 2 mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,6 +182,27 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + description: The phandle of an S/PDIF CPU DAI controller. + + spdif-out: + type: boolean + deprecated: true + description: | + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + deprecated: true + description: | + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + required: - compatible - model @@ -195,3 +231,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; -- 2.34.1

2 1

[PATCH v1] ASoc: TAS2781: replace beXX_to_cpup with get_unaligned_beXX for potentially broken alignment
by Shenghao Ding 08 Jul '24

08 Jul '24

Use get_unaligned_be16 instead of be16_to_cpup and get_unaligned_be32 instead of be32_to_cpup for potentially broken alignment. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add "#include <asm/unaligned.h>" for the declaration of get_unaligned_beXX --- sound/soc/codecs/tas2781-fmwlib.c | 71 +++++++++++++++---------------- 1 file changed, 35 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 838d29fead96..63626b982d04 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -21,7 +21,7 @@ #include <sound/soc.h> #include <sound/tlv.h> #include <sound/tas2781.h> - +#include <asm/unaligned.h> #define ERROR_PRAM_CRCCHK 0x0000000 #define ERROR_YRAM_CRCCHK 0x0000001 @@ -187,8 +187,7 @@ static struct tasdevice_config_info *tasdevice_add_config( /* convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - cfg_info->nblocks = - be32_to_cpup((__be32 *)&config_data[config_offset]); + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); config_offset += 4; /* Several kinds of dsp/algorithm firmwares can run on tas2781, @@ -232,14 +231,14 @@ static struct tasdevice_config_info *tasdevice_add_config( } bk_da[i]->yram_checksum = - be16_to_cpup((__be16 *)&config_data[config_offset]); + get_unaligned_be16(&config_data[config_offset]); config_offset += 2; bk_da[i]->block_size = - be32_to_cpup((__be32 *)&config_data[config_offset]); + get_unaligned_be32(&config_data[config_offset]); config_offset += 4; bk_da[i]->n_subblks = - be32_to_cpup((__be32 *)&config_data[config_offset]); + get_unaligned_be32(&config_data[config_offset]); config_offset += 4; @@ -289,7 +288,7 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) } buf = (unsigned char *)fmw->data; - fw_hdr->img_sz = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); offset += 4; if (fw_hdr->img_sz != fmw->size) { dev_err(tas_priv->dev, @@ -300,9 +299,9 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) goto out; } - fw_hdr->checksum = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); offset += 4; - fw_hdr->binary_version_num = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); if (fw_hdr->binary_version_num < 0x103) { dev_err(tas_priv->dev, "File version 0x%04x is too low", fw_hdr->binary_version_num); @@ -311,7 +310,7 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) goto out; } offset += 4; - fw_hdr->drv_fw_version = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); offset += 8; fw_hdr->plat_type = buf[offset]; offset += 1; @@ -339,11 +338,11 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) fw_hdr->devs[i] = buf[offset]; - fw_hdr->nconfig = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); offset += 4; for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { - fw_hdr->config_size[i] = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); offset += 4; total_config_sz += fw_hdr->config_size[i]; } @@ -423,7 +422,7 @@ static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, /* convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - block->type = be32_to_cpup((__be32 *)&data[offset]); + block->type = get_unaligned_be32(&data[offset]); offset += 4; block->is_pchksum_present = data[offset]; @@ -438,10 +437,10 @@ static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, block->ychksum = data[offset]; offset++; - block->blk_size = be32_to_cpup((__be32 *)&data[offset]); + block->blk_size = get_unaligned_be32(&data[offset]); offset += 4; - block->nr_subblocks = be32_to_cpup((__be32 *)&data[offset]); + block->nr_subblocks = get_unaligned_be32(&data[offset]); offset += 4; /* fixed m68k compiling issue: @@ -482,7 +481,7 @@ static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, offset = -EINVAL; goto out; } - img_data->nr_blk = be32_to_cpup((__be32 *)&data[offset]); + img_data->nr_blk = get_unaligned_be32(&data[offset]); offset += 4; img_data->dev_blks = kcalloc(img_data->nr_blk, @@ -578,14 +577,14 @@ static int fw_parse_variable_header_kernel( offset = -EINVAL; goto out; } - fw_hdr->device_family = be16_to_cpup((__be16 *)&buf[offset]); + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); if (fw_hdr->device_family != 0) { dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); offset = -EINVAL; goto out; } offset += 2; - fw_hdr->device = be16_to_cpup((__be16 *)&buf[offset]); + fw_hdr->device = get_unaligned_be16(&buf[offset]); if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || fw_hdr->device == 6) { dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); @@ -603,7 +602,7 @@ static int fw_parse_variable_header_kernel( goto out; } - tas_fmw->nr_programs = be32_to_cpup((__be32 *)&buf[offset]); + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); offset += 4; if (tas_fmw->nr_programs == 0 || tas_fmw->nr_programs > @@ -622,14 +621,14 @@ static int fw_parse_variable_header_kernel( for (i = 0; i < tas_fmw->nr_programs; i++) { program = &(tas_fmw->programs[i]); - program->prog_size = be32_to_cpup((__be32 *)&buf[offset]); + program->prog_size = get_unaligned_be32(&buf[offset]); offset += 4; } /* Skip the unused prog_size */ offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); - tas_fmw->nr_configurations = be32_to_cpup((__be32 *)&buf[offset]); + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); offset += 4; /* The max number of config in firmware greater than 4 pieces of @@ -661,7 +660,7 @@ static int fw_parse_variable_header_kernel( for (i = 0; i < tas_fmw->nr_programs; i++) { config = &(tas_fmw->configs[i]); - config->cfg_size = be32_to_cpup((__be32 *)&buf[offset]); + config->cfg_size = get_unaligned_be32(&buf[offset]); offset += 4; } @@ -699,7 +698,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, switch (subblk_typ) { case TASDEVICE_CMD_SING_W: { int i; - unsigned short len = be16_to_cpup((__be16 *)&data[2]); + unsigned short len = get_unaligned_be16(&data[2]); subblk_offset += 2; if (subblk_offset + 4 * len > sublocksize) { @@ -725,7 +724,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, } break; case TASDEVICE_CMD_BURST: { - unsigned short len = be16_to_cpup((__be16 *)&data[2]); + unsigned short len = get_unaligned_be16(&data[2]); subblk_offset += 2; if (subblk_offset + 4 + len > sublocksize) { @@ -766,7 +765,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, is_err = true; break; } - sleep_time = be16_to_cpup((__be16 *)&data[2]) * 1000; + sleep_time = get_unaligned_be16(&data[2]) * 1000; usleep_range(sleep_time, sleep_time + 50); subblk_offset += 2; } @@ -910,7 +909,7 @@ static int fw_parse_variable_hdr(struct tasdevice_priv offset += len; - fw_hdr->device_family = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); if (fw_hdr->device_family != 0) { dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); offset = -EINVAL; @@ -918,7 +917,7 @@ static int fw_parse_variable_hdr(struct tasdevice_priv } offset += 4; - fw_hdr->device = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->device = get_unaligned_be32(&buf[offset]); if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || fw_hdr->device == 6) { dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); @@ -963,7 +962,7 @@ static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, offset = -EINVAL; goto out; } - block->type = be32_to_cpup((__be32 *)&data[offset]); + block->type = get_unaligned_be32(&data[offset]); offset += 4; if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { @@ -988,7 +987,7 @@ static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, block->is_ychksum_present = 0; } - block->nr_cmds = be32_to_cpup((__be32 *)&data[offset]); + block->nr_cmds = get_unaligned_be32(&data[offset]); offset += 4; n = block->nr_cmds * 4; @@ -1039,7 +1038,7 @@ static int fw_parse_data(struct tasdevice_fw *tas_fmw, goto out; } offset += n; - img_data->nr_blk = be16_to_cpup((__be16 *)&data[offset]); + img_data->nr_blk = get_unaligned_be16(&data[offset]); offset += 2; img_data->dev_blks = kcalloc(img_data->nr_blk, @@ -1076,7 +1075,7 @@ static int fw_parse_program_data(struct tasdevice_priv *tas_priv, offset = -EINVAL; goto out; } - tas_fmw->nr_programs = be16_to_cpup((__be16 *)&buf[offset]); + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); offset += 2; if (tas_fmw->nr_programs == 0) { @@ -1143,7 +1142,7 @@ static int fw_parse_configuration_data( offset = -EINVAL; goto out; } - tas_fmw->nr_configurations = be16_to_cpup((__be16 *)&data[offset]); + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); offset += 2; if (tas_fmw->nr_configurations == 0) { @@ -1775,7 +1774,7 @@ static int fw_parse_header(struct tasdevice_priv *tas_priv, /* Convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - fw_fixed_hdr->fwsize = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); offset += 4; if (fw_fixed_hdr->fwsize != fmw->size) { dev_err(tas_priv->dev, "File size not match, %lu %u", @@ -1784,9 +1783,9 @@ static int fw_parse_header(struct tasdevice_priv *tas_priv, goto out; } offset += 4; - fw_fixed_hdr->ppcver = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); offset += 8; - fw_fixed_hdr->drv_ver = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); offset += 72; out: @@ -1828,7 +1827,7 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, offset = -EINVAL; goto out; } - tas_fmw->nr_calibrations = be16_to_cpup((__be16 *)&data[offset]); + tas_fmw->nr_calibrations = get_unaligned_be16(&data[offset]); offset += 2; if (tas_fmw->nr_calibrations != 1) { -- 2.34.1

2 1

[PATCH alsa-cum-conf v3 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

1 1

Numark Scratch Mixer USB Audio Interface
by amhadebe＠gmail.com 08 Jul '24

08 Jul '24

Hi All, I have a Numark Scratch mixer which I would like to use on Linux with Mixxx. The mixer should be class compliant but there is an issue and the Audio interface does not work. I hope to get your help in getting this fixed in the kernel. I'm on fedora 49 running kernel 6.9.5-200 Here's some info, I have found: $ dmesg [ 74.473642] usb 1-3: new high-speed USB device number 5 using xhci_hcd [ 74.600370] usb 1-3: language id specifier not provided by device, defaulting to English [ 74.600541] usb 1-3: New USB device found, idVendor=15e4, idProduct=0052, bcdDevice= 0.00 [ 74.600556] usb 1-3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.600566] usb 1-3: Product: Numark Scratch [ 74.600574] usb 1-3: Manufacturer: Numark [ 74.600580] usb 1-3: SerialNumber: N31911167105329 [ 74.604272] usb 1-3: Quirk or no altset; falling back to MIDI 1.0 [ 74.606706] usb 1-3: 3:1 : bogus bTerminalLink 1 [ 74.606802] usb 1-3: 4:1 : bogus bTerminalLink 4 [ 156.432125] usb 1-3: USB disconnect, device number 5 $ amidi -l Dir Device Name IO hw:0,0,0 Numark Scratch MIDI 1 $ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: CX20753/4 Analog [CX20753/4 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] $ lsusb -v Bus 001 Device 045: ID 15e4:0052 Numark Numark Scratch Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 [unknown] bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x15e4 Numark idProduct 0x0052 Numark Scratch bcdDevice 0.00 iManufacturer 1 Numark iProduct 2 Numark Scratch iSerial 3 N31911167105329 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0164 bNumInterfaces 5 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0009 bInCollection 1 baInterfaceNr(0) 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0032 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 2 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (Invalid) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (Invalid) bNumEmbMIDIJack 1 baAssocJackID( 0) 2 Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 32 iFunction 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0087 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 5 bmAttributes 1 Internal fixed clock bmControls 0x05 Clock Frequency Control (read-only) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 2 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 5 bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 4 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 1 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 bmaControls(3) 0x00000000 bmaControls(4) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 6 bCSourceID 5 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 5 bNrChannels 6 bmChannelConfig 0x00000000 iChannelNames 8 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 7 bSourceID 3 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 bmaControls(3) 0x00000000 bmaControls(4) 0x00000000 bmaControls(5) 0x00000000 bmaControls(6) 0x00000000 iFeature 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 7 bCSourceID 5 bmControls 0x0000 iTerminal 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 4 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 6 bmChannelConfig 0x00000000 iChannelNames 8 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x00a8 1x 168 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Regards Al

1 0

Fix labeling of BOSS Katana ports
by GitHub pull_request - opened 06 Jul '24

06 Jul '24

alsa-project/alsa-ucm-conf pull request #430 was opened from aledomu: I'd like to rename the MIDI ports as well but I don't know how. If you can point me where I can learn about it, I'll be glad to do so and mark this PR as finished. I guess this fix works the same for all Katana models from all generations, but I don't have the PID available to enable it, nor the devices for testing. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-utils 1/5] aplaymidi2: Add initial version
by Takashi Iwai 06 Jul '24

06 Jul '24

aplaymidi2 is a program similar like aplaymidi, but intended for playing back a MIDI Clip file that was introduced for handling UMP. MIDI Clip file contains UMP packets, and its structure is much simpler than SMF. The options are mostly same as aplaymidi, but I omitted -l option for simplifying the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 2 +- seq/Makefile.am | 2 +- seq/aplaymidi2/Makefile.am | 5 + seq/aplaymidi2/aplaymidi2.1 | 71 ++++++ seq/aplaymidi2/aplaymidi2.c | 447 ++++++++++++++++++++++++++++++++++++ 5 files changed, 525 insertions(+), 2 deletions(-) create mode 100644 seq/aplaymidi2/Makefile.am create mode 100644 seq/aplaymidi2/aplaymidi2.1 create mode 100644 seq/aplaymidi2/aplaymidi2.c diff --git a/configure.ac b/configure.ac index 173b9ed1644c..708380cf31dc 100644 --- a/configure.ac +++ b/configure.ac @@ -486,7 +486,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ bat/Makefile bat/tests/Makefile bat/tests/asound_state/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ - seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ + seq/aplaymidi/Makefile seq/aplaymidi2/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ seq/aseqsend/Makefile speaker-test/Makefile speaker-test/samples/Makefile \ alsaloop/Makefile alsa-info/Makefile \ axfer/Makefile axfer/test/Makefile \ diff --git a/seq/Makefile.am b/seq/Makefile.am index b0f628aac4da..a6171268186b 100644 --- a/seq/Makefile.am +++ b/seq/Makefile.am @@ -1 +1 @@ -SUBDIRS=aconnect aplaymidi aseqdump aseqnet aseqsend +SUBDIRS=aconnect aplaymidi aplaymidi2 aseqdump aseqnet aseqsend diff --git a/seq/aplaymidi2/Makefile.am b/seq/aplaymidi2/Makefile.am new file mode 100644 index 000000000000..0c5c743f7eb8 --- /dev/null +++ b/seq/aplaymidi2/Makefile.am @@ -0,0 +1,5 @@ +AM_CPPFLAGS = -I$(top_srcdir)/include +EXTRA_DIST = aplaymidi2.1 + +bin_PROGRAMS = aplaymidi2 +man_MANS = aplaymidi2.1 diff --git a/seq/aplaymidi2/aplaymidi2.1 b/seq/aplaymidi2/aplaymidi2.1 new file mode 100644 index 000000000000..99be608f107d --- /dev/null +++ b/seq/aplaymidi2/aplaymidi2.1 @@ -0,0 +1,71 @@ +.TH APLAYMIDI2 1 "4 July 2024" + +.SH NAME +aplaymidi2 \- play MIDI Clip Files + +.SH SYNOPSIS +.B aplaymidi2 +\-p client:port[,...] midi2file ... + +.SH DESCRIPTION +.B aplaymidi2 +is a command-line utility that plays the specified MIDI Clip file(s) to one +or more ALSA sequencer ports. + +.SH OPTIONS + +.TP +.I \-h, \-\-help +Prints a list of options. + +.TP +.I \-V, \-\-version +Prints the current version. + +.TP +.I \-p, \-\-port=client:port,... +Sets the sequencer port(s) to which the events in the MIDI Clip file(s) are +sent. + +A client can be specified by its number, its name, or a prefix of its +name. A port is specified by its number; for port 0 of a client, the +":0" part of the port specification can be omitted. + +Multiple ports can be specified to allow playback of MIDI Clip file(s) that +contain events for multiple devices (ports) corresponding to the +multiple UMP Groups. + +For compatibility with +.B pmidi(1), +the port specification is taken from the +.I ALSA_OUTPUT_PORTS +environment variable if none is given on the command line. + +.B aplaymidi2 +supports only basic UMP events: in addition to the standard MIDI1 and +MIDI2 CVMs and 7bit SysEx, only the following are supported: +DCTPQ, DC, Set Tempo, Start Clip and End Clip. +Lyrics and other meta data in Flex Data are skipped, so far. + +The multiple output ports are useful when the given MIDI Clip file +contains the UMP packets for multiple Groups. +When the destination port is a UMP MIDI 2.0 port, the single +connection should suffice, though, since a MIDI 2.0 port can process +the inputs for multiple Groups. For other cases (e.g. connecting to a +legacy MIDI port), you would need to specify the destination port per +Group. If undefined, it's sent to the first destination port as +default. + +.TP +.I \-d, \-\-delay=seconds +Specifies how long to wait after the end of each MIDI Clip file, +to allow the last notes to die away. +Default is 2 seconds. + +.SH SEE ALSO +pmidi(1) +.br +aplaymidi(1) + +.SH AUTHOR +Takashi Iwai <tiwai(a)suse.de> diff --git a/seq/aplaymidi2/aplaymidi2.c b/seq/aplaymidi2/aplaymidi2.c new file mode 100644 index 000000000000..435b7de1aa95 --- /dev/null +++ b/seq/aplaymidi2/aplaymidi2.c @@ -0,0 +1,447 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * aplaymidi2.c - simple player of a MIDI Clip File over ALSA sequencer + */ + +#include "aconfig.h" +#include <stdio.h> +#include <stdlib.h> +#include <stdarg.h> +#include <string.h> +#include <getopt.h> +#include <unistd.h> +#include <alsa/asoundlib.h> +#include <alsa/ump_msg.h> +#include "version.h" + +static snd_seq_t *seq; +static int client; +static int port_count; +static snd_seq_addr_t ports[16]; +static int queue; +static int end_delay = 2; + +static unsigned int _current_tempo = 50000000; /* default 120 bpm */ +static unsigned int tempo_base = 10; +static unsigned int current_tick; + +/* prints an error message to stderr */ +static void errormsg(const char *msg, ...) +{ + va_list ap; + + va_start(ap, msg); + vfprintf(stderr, msg, ap); + va_end(ap); + fputc('\n', stderr); +} + +/* prints an error message to stderr, and dies */ +static void fatal(const char *msg, ...) +{ + va_list ap; + + va_start(ap, msg); + vfprintf(stderr, msg, ap); + va_end(ap); + fputc('\n', stderr); + exit(EXIT_FAILURE); +} + +/* memory allocation error handling */ +static void check_mem(void *p) +{ + if (!p) + fatal("Out of memory"); +} + +/* error handling for ALSA functions */ +static void check_snd(const char *operation, int err) +{ + if (err < 0) + fatal("Cannot %s - %s", operation, snd_strerror(err)); +} + +/* open and initialize the sequencer client */ +static void init_seq(void) +{ + int err; + + err = snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); + check_snd("open sequencer", err); + + err = snd_seq_set_client_name(seq, "aplaymidi2"); + check_snd("set client name", err); + + client = snd_seq_client_id(seq); + check_snd("get client id", client); + + err = snd_seq_set_client_midi_version(seq, SND_SEQ_CLIENT_UMP_MIDI_2_0); + check_snd("set midi version", err); +} + +/* parses one or more port addresses from the string */ +static void parse_ports(const char *arg) +{ + char *buf, *s, *port_name; + int err; + + /* make a copy of the string because we're going to modify it */ + buf = strdup(arg); + check_mem(buf); + + for (port_name = s = buf; s; port_name = s + 1) { + /* Assume that ports are separated by commas. We don't use + * spaces because those are valid in client names. */ + s = strchr(port_name, ','); + if (s) + *s = '\0'; + + ++port_count; + if (port_count > 16) + fatal("Too many ports specified"); + + err = snd_seq_parse_address(seq, &ports[port_count - 1], port_name); + if (err < 0) + fatal("Invalid port %s - %s", port_name, snd_strerror(err)); + } + + free(buf); +} + +/* create a source port to send from */ +static void create_source_port(void) +{ + snd_seq_port_info_t *pinfo; + int err; + + snd_seq_port_info_alloca(&pinfo); + + /* the first created port is 0 anyway, but let's make sure ... */ + snd_seq_port_info_set_port(pinfo, 0); + snd_seq_port_info_set_port_specified(pinfo, 1); + + snd_seq_port_info_set_name(pinfo, "aplaymidi2"); + + snd_seq_port_info_set_capability(pinfo, 0); /* sic */ + snd_seq_port_info_set_type(pinfo, + SND_SEQ_PORT_TYPE_MIDI_GENERIC | + SND_SEQ_PORT_TYPE_APPLICATION); + + err = snd_seq_create_port(seq, pinfo); + check_snd("create port", err); +} + +/* create a queue */ +static void create_queue(void) +{ + if (!snd_seq_has_queue_tempo_base(seq)) + tempo_base = 1000; + + queue = snd_seq_alloc_named_queue(seq, "aplaymidi2"); + check_snd("create queue", queue); +} + +/* connect to destination ports */ +static void connect_ports(void) +{ + int i, err; + + for (i = 0; i < port_count; ++i) { + err = snd_seq_connect_to(seq, 0, ports[i].client, ports[i].port); + if (err < 0) + fatal("Cannot connect to port %d:%d - %s", + ports[i].client, ports[i].port, snd_strerror(err)); + } +} + +/* read 32bit word and convert to native endian: + * return 0 on success, -1 on error + */ +static int read_word(FILE *file, uint32_t *dest) +{ + uint32_t v; + + if (fread(&v, 4, 1, file) != 1) + return -1; + *dest = be32toh(v); + return 0; +} + +/* read a UMP packet: return the number of packets, -1 on error */ +static int read_ump_packet(FILE *file, uint32_t *buf) +{ + snd_ump_msg_hdr_t *h = (snd_ump_msg_hdr_t *)buf; + + int i, num; + + if (read_word(file, buf) < 0) + return -1; + num = snd_ump_packet_length(h->type); + for (i = 1; i < num; i++) { + if (read_word(file, buf + i) < 0) + return -1; + } + return num; +} + +/* read the file header and verify it's MIDI Clip File: return 0 on success */ +static int verify_file_header(FILE *file) +{ + unsigned char buf[8]; + + if (fread(buf, 1, 8, file) != 8) + return -1; + if (memcmp(buf, "SMF2CLIP", 8)) + return -1; + return 0; +} + +/* return the current tempo, corrected to be sent to host */ +static int current_tempo(void) +{ + if (tempo_base != 10) + return _current_tempo / 100; /* down to us */ + return _current_tempo; +} + +/* send a timer event */ +static void send_timer_event(unsigned int type, unsigned int val) +{ + snd_seq_ump_event_t ev = { + .type = type, + .flags = SND_SEQ_TIME_STAMP_TICK | SND_SEQ_EVENT_LENGTH_FIXED, + }; + + ev.queue = queue; + ev.source.port = 0; + ev.time.tick = current_tick; + + ev.dest.client = SND_SEQ_CLIENT_SYSTEM; + ev.dest.port = SND_SEQ_PORT_SYSTEM_TIMER; + ev.data.queue.queue = queue; + ev.data.queue.param.value = val; + + snd_seq_ump_event_output(seq, &ev); +} + +/* set DCTPQ */ +static void set_dctpq(unsigned int ppq) +{ + snd_seq_queue_tempo_t *queue_tempo; + + snd_seq_queue_tempo_alloca(&queue_tempo); + snd_seq_queue_tempo_set_tempo(queue_tempo, current_tempo()); + snd_seq_queue_tempo_set_ppq(queue_tempo, ppq); + snd_seq_queue_tempo_set_tempo_base(queue_tempo, tempo_base); + + if (snd_seq_set_queue_tempo(seq, queue, queue_tempo) < 0) + errormsg("Cannot set queue tempo (%d)", queue); +} + +/* set DC */ +static void set_dc(unsigned int ticks) +{ + current_tick += ticks; +} + +/* set tempo event */ +static void set_tempo(unsigned int tempo) +{ + _current_tempo = tempo; + send_timer_event(SND_SEQ_EVENT_TEMPO, current_tempo()); +} + +/* start clip */ +static void start_clip(void) +{ + if (snd_seq_start_queue(seq, queue, NULL) < 0) + errormsg("Cannot start queue (%d)", queue); +} + +/* end clip */ +static void end_clip(void) +{ + send_timer_event(SND_SEQ_EVENT_STOP, 0); +} + +/* send a UMP packet */ +static void send_ump(const uint32_t *ump, int len) +{ + snd_seq_ump_event_t ev = { + .flags = SND_SEQ_TIME_STAMP_TICK | SND_SEQ_EVENT_LENGTH_FIXED | + SND_SEQ_EVENT_UMP, + }; + int group; + + memcpy(ev.ump, ump, len * 4); + + ev.queue = queue; + ev.source.port = 0; + ev.time.tick = current_tick; + group = snd_ump_msg_group(ump); + if (group >= port_count) + ev.dest = ports[0]; + else + ev.dest = ports[group]; + + snd_seq_ump_event_output(seq, &ev); +} + +/* play the given MIDI Clip File content */ +static void play_midi(FILE *file) +{ + uint32_t ump[4]; + int len; + + current_tick = 0; + + while ((len = read_ump_packet(file, ump)) > 0) { + const snd_ump_msg_hdr_t *h = (snd_ump_msg_hdr_t *)ump; + + if (h->type == SND_UMP_MSG_TYPE_UTILITY) { + const snd_ump_msg_utility_t *uh = + (const snd_ump_msg_utility_t *)ump; + switch (h->status) { + case SND_UMP_UTILITY_MSG_STATUS_DCTPQ: + set_dctpq(uh->dctpq.ticks); + continue; + case SND_UMP_UTILITY_MSG_STATUS_DC: + set_dc(uh->dctpq.ticks); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_FLEX_DATA) { + const snd_ump_msg_flex_data_t *fh = + (const snd_ump_msg_flex_data_t *)ump; + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_TEMPO) { + set_tempo(fh->set_tempo.tempo); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_STREAM) { + const snd_ump_msg_stream_t *sh = + (const snd_ump_msg_stream_t *)ump; + switch (sh->gen.status) { + case SND_UMP_STREAM_MSG_STATUS_START_CLIP: + start_clip(); + continue; + case SND_UMP_STREAM_MSG_STATUS_END_CLIP: + end_clip(); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE || + h->type == SND_UMP_MSG_TYPE_DATA || + h->type == SND_UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE) { + send_ump(ump, len); + } + } + + snd_seq_drain_output(seq); + snd_seq_sync_output_queue(seq); + + /* give the last notes time to die away */ + if (end_delay > 0) + sleep(end_delay); +} + +static void play_file(const char *file_name) +{ + FILE *file; + + if (!strcmp(file_name, "-")) + file = stdin; + else + file = fopen(file_name, "rb"); + if (!file) { + errormsg("Cannot open %s - %s", file_name, strerror(errno)); + return; + } + + if (verify_file_header(file) < 0) { + errormsg("%s is not a MIDI Clip File", file_name); + goto error; + } + + play_midi(file); + + error: + if (file != stdin) + fclose(file); +} + +static void usage(const char *argv0) +{ + printf( + "Usage: %s -p client:port[,...] [-d delay] midifile ...\n" + "-h, --help this help\n" + "-V, --version print current version\n" + "-p, --port=client:port,... set port(s) to play to\n" + "-d, --delay=seconds delay after song ends\n", + argv0); +} + +static void version(void) +{ + puts("aplaymidi2 version " SND_UTIL_VERSION_STR); +} + +int main(int argc, char *argv[]) +{ + static const struct option long_options[] = { + {"help", 0, NULL, 'h'}, + {"version", 0, NULL, 'V'}, + {"port", 1, NULL, 'p'}, + {"delay", 1, NULL, 'd'}, + {0} + }; + int c; + + init_seq(); + + while ((c = getopt_long(argc, argv, "hVp:d:", + long_options, NULL)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'V': + version(); + return 0; + case 'p': + parse_ports(optarg); + break; + case 'd': + end_delay = atoi(optarg); + break; + default: + usage(argv[0]); + return 1; + } + } + + + if (port_count < 1) { + /* use env var for compatibility with pmidi */ + const char *ports_str = getenv("ALSA_OUTPUT_PORTS"); + if (ports_str) + parse_ports(ports_str); + if (port_count < 1) { + errormsg("Please specify at least one port with --port."); + return 1; + } + } + if (optind >= argc) { + errormsg("Please specify a file to play."); + return 1; + } + + create_source_port(); + create_queue(); + connect_ports(); + + for (; optind < argc; optind++) + play_file(argv[optind]); + + snd_seq_close(seq); + return 0; +} -- 2.43.0

1 4

[PATCH alsa-lib] seq: Add API functions to set different tempo base values
by Takashi Iwai 06 Jul '24

06 Jul '24

MIDI2 Set Tempo message uses 10ns-based values, and we need to update the API to change the base time unit. This patch adds a few new API functions: - snd_seq_has_queue_tempo_base() returns 1 if the client supports a new tempo-base value; if 0, it's an old system and application has to use the tempo in the fixed 1us unit - the tempo base can be changed with snd_seq_queue_tempo_set_tempo_base(), provided in nsec unit; the value has to be either 10 or 1000 (or 0 as default, equivalent with 1000) The protocol version is checked and fallback to the fixed 1us base for the old clients. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 3 ++ include/sound/uapi/asequencer.h | 7 ++-- src/Versions.in.in | 3 ++ src/seq/seq.c | 58 +++++++++++++++++++++++++++++++++ src/seq/seq_hw.c | 5 +++ src/seq/seq_local.h | 1 + 6 files changed, 74 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index e55f5c16fada..3bad5dd9762c 100644 --- a/include/seq.h +++ b/include/seq.h @@ -506,13 +506,16 @@ unsigned int snd_seq_queue_tempo_get_tempo(const snd_seq_queue_tempo_t *info); int snd_seq_queue_tempo_get_ppq(const snd_seq_queue_tempo_t *info); unsigned int snd_seq_queue_tempo_get_skew(const snd_seq_queue_tempo_t *info); unsigned int snd_seq_queue_tempo_get_skew_base(const snd_seq_queue_tempo_t *info); +unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t *info); void snd_seq_queue_tempo_set_tempo(snd_seq_queue_tempo_t *info, unsigned int tempo); void snd_seq_queue_tempo_set_ppq(snd_seq_queue_tempo_t *info, int ppq); void snd_seq_queue_tempo_set_skew(snd_seq_queue_tempo_t *info, unsigned int skew); void snd_seq_queue_tempo_set_skew_base(snd_seq_queue_tempo_t *info, unsigned int base); +void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t *info, unsigned int tempo_base); int snd_seq_get_queue_tempo(snd_seq_t *handle, int q, snd_seq_queue_tempo_t *tempo); int snd_seq_set_queue_tempo(snd_seq_t *handle, int q, snd_seq_queue_tempo_t *tempo); +int snd_seq_has_queue_tempo_base(snd_seq_t *handle); /* */ diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index b913f31daa2d..923dfdddfc76 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -26,7 +26,7 @@ #include <sound/asound.h> /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 3) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 4) /** * definition of sequencer event types @@ -539,11 +539,12 @@ struct snd_seq_queue_status { /* queue tempo */ struct snd_seq_queue_tempo { int queue; /* sequencer queue */ - unsigned int tempo; /* current tempo, us/tick */ + unsigned int tempo; /* current tempo, us/tick (or different time-base below) */ int ppq; /* time resolution, ticks/quarter */ unsigned int skew_value; /* queue skew */ unsigned int skew_base; /* queue skew base */ - char reserved[24]; /* for the future */ + unsigned short tempo_base; /* tempo base in nsec unit; either 10 or 1000 */ + char reserved[22]; /* for the future */ }; diff --git a/src/Versions.in.in b/src/Versions.in.in index 1c40d461dce4..90849277c983 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -206,6 +206,9 @@ ALSA_1.2.13 { #ifdef HAVE_SEQ_SYMS @SYMBOL_PREFIX@snd_seq_create_ump_endpoint; @SYMBOL_PREFIX@snd_seq_create_ump_block; + @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; + @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; + @SYMBOL_PREFIX@snd_seq_has_tempo_base; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index a18f67e2632e..d0ecedf0838f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -490,6 +490,21 @@ the buffer is flushed by #snd_seq_drain_output() call. You can schedule the event in a certain queue so that the tempo change happens at the scheduled time, too. +The tempo is set as default in microsecond unit as defined for +Standard MIDI Format 1. But since the value in MIDI2 Set Tempo message +is based on 10-nanosecand unit, sequencer queue also allows to set up +in 10-nanosecond unit. For that, change the tempo-base value in +#snd_seq_queue_tempo_t to 10 via #snd_seq_queue_tempo_set_tempo_base() +along with the 10-nanobased tempo value. The default tempo base is 1000, +i.e. 1 microsecond. +Currently the API supports only either 0, 10 or 1000 as the tempo-base +(where 0 is treated as 1000). + +The older kernel might not support the different tempo-base, and setting a +different value from 1000 would fail. The application may heck the +availability of tempo-base change via #snd_seq_has_tempo_base() function +beforehand, and re-calculate in microsecond unit as fallback. + \subsection seq_ev_start Starting and stopping a queue To start, stop, or continue a queue, you need to send a queue-control @@ -3878,6 +3893,19 @@ unsigned int snd_seq_queue_tempo_get_skew_base(const snd_seq_queue_tempo_t *info return info->skew_base; } +/** + * \brief Get the tempo base of a queue_status container + * \param info queue_status container + * \return tempo base time in nsec unit + * + * \sa snd_seq_get_queue_tempo() + */ +unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t *info) +{ + assert(info); + return info->tempo_base; +} + /** * \brief Set the tempo of a queue_status container * \param info queue_status container @@ -3933,6 +3961,21 @@ void snd_seq_queue_tempo_set_skew_base(snd_seq_queue_tempo_t *info, unsigned int info->skew_base = base; } +/** + * \brief Set the tempo base of a queue_status container + * \param info queue_status container + * \param tempo_base tempo base time in nsec unit + * + * \sa snd_seq_get_queue_tempo() + */ +void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t *info, unsigned int tempo_base) +{ + assert(info); + if (!tempo_base) + tempo_base = 1000; + info->tempo_base = tempo_base; +} + /** * \brief obtain the current tempo of the queue * \param seq sequencer handle @@ -3962,10 +4005,25 @@ int snd_seq_get_queue_tempo(snd_seq_t *seq, int q, snd_seq_queue_tempo_t * tempo int snd_seq_set_queue_tempo(snd_seq_t *seq, int q, snd_seq_queue_tempo_t * tempo) { assert(seq && tempo); + if (!seq->has_queue_tempo_base && + tempo->tempo_base && tempo->tempo_base != 1000) + return -EINVAL; tempo->queue = q; return seq->ops->set_queue_tempo(seq, tempo); } +/** + * \brief inquiry the support of tempo base change + * \param seq sequencer handle + * \return 1 if the client supports the tempo base change, 0 if not + * + * \sa snd_seq_get_queue_tempo() + */ +int snd_seq_has_queue_tempo_base(snd_seq_t *seq) +{ + assert(seq); + return seq->has_queue_tempo_base; +} /*----------------------------------------------------------------*/ diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index eeaf26e16d1c..e88a7b22b052 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -275,12 +275,15 @@ static int snd_seq_hw_get_queue_tempo(snd_seq_t *seq, snd_seq_queue_tempo_t * te /*SYSERR("SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO failed");*/ return -errno; } + if (!seq->has_queue_tempo_base) + tempo->tempo_base = 1000; return 0; } static int snd_seq_hw_set_queue_tempo(snd_seq_t *seq, snd_seq_queue_tempo_t * tempo) { snd_seq_hw_t *hw = seq->private_data; + if (ioctl(hw->fd, SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO, tempo) < 0) { /*SYSERR("SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO failed");*/ return -errno; @@ -587,6 +590,8 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) seq->ops = &snd_seq_hw_ops; seq->private_data = hw; seq->packet_size = sizeof(snd_seq_event_t); + seq->has_queue_tempo_base = ver >= SNDRV_PROTOCOL_VERSION(1, 0, 4); + client = snd_seq_hw_client_id(seq); if (client < 0) { snd_seq_close(seq); diff --git a/src/seq/seq_local.h b/src/seq/seq_local.h index 263029702739..00a7615afeae 100644 --- a/src/seq/seq_local.h +++ b/src/seq/seq_local.h @@ -94,6 +94,7 @@ struct _snd_seq { size_t tmpbufsize; /* size of errbuf */ size_t packet_size; /* input packet alignment size */ int midi_version; /* current protocol version */ + int has_queue_tempo_base; /* support queue tempo-base? */ unsigned int num_ump_groups; /* number of UMP groups */ snd_ump_endpoint_info_t *ump_ep; /* optional UMP info */ -- 2.43.0

1 0

Add support for a new DAC
by Pino Abete 05 Jul '24

05 Jul '24

Hi, I'm not an expert and I'm sorry if this is the wrong place for thsi kind of requests. I'm trying to add native DSD support for this integrated DAC/amplifier: https://www.mission.co.uk/778x/ Unfortunately, vendor is not collaborative. It should support up to 384kHz (PCM) and DSD256 and have a XMOS USB chip and a ESS Sabre32 ES9018K2M DAC chip. I tried simply to add: case USB_ID(0x2622, 0x0011): /* Mission 778X */ in the "switch" section under /* XMOS based USB DACs */ and it works with the DSD_U32_BE format (tried using MPD - Music Player Demo). Is it possible to officially add this to source or do you need a request from OEM? Thanks --

1 0

Re: [PATCH] Fix swapped l/r audio channels for Lenovo ThinBook 13x Gen4
by Takashi Iwai 05 Jul '24

05 Jul '24

On Thu, 04 Jul 2024 23:14:01 +0200, Dmitry Savin wrote: > > Fixes audio channel assignment in configuration table for ThinkBook 13x Gen4. > > Signed-off-by: Dmitry Savin <envelsavinds(a)gmail.com> > Reviewed-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Thanks, applied now. Takashi

1 0

[tiwai-sound:for-next 5/6] ERROR: modpost: "snd_pcm_set_sync_per_card" [sound/pci/emu10k1/snd-emu10k1.ko] undefined!
by kernel test robot 05 Jul '24

05 Jul '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 5f9f982dd71b418aeba7a0b37f87312545c06df4 commit: d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 [5/6] ALSA: pcm: optimize and clarify stream synchronization ID API config: i386-randconfig-r064-20240705 (https://download.01.org/0day-ci/archive/20240705/202407051343.1Sz8AKFt-lkp@…) compiler: gcc-11 (Ubuntu 11.4.0-4ubuntu1) 11.4.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240705/202407051343.1Sz8AKFt-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202407051343.1Sz8AKFt-lkp@intel.com/ All errors (new ones prefixed by >>, old ones prefixed by <<): WARNING: modpost: missing MODULE_DESCRIPTION() in vmlinux.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/locking/locktorture.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/locking/test-ww_mutex.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/rcutorture.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/rcuscale.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/refscale.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/time/test_udelay.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/time/clocksource-wdtest.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/backtracetest.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/torture.o WARNING: modpost: missing MODULE_DESCRIPTION() in mm/dmapool_test.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nfs/nfsv4.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp737.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp855.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp863.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp866.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp936.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp950.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_ascii.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_iso8859-7.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_iso8859-13.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-r.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-u.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-ru.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-croatian.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-inuit.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-romanian.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-turkish.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_ucs2_utils.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/binfmt_misc.o WARNING: modpost: missing MODULE_DESCRIPTION() in security/keys/encrypted-keys/encrypted-keys.o WARNING: modpost: missing MODULE_DESCRIPTION() in crypto/ecc.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/crypto/libdes.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_hexdump.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_dhry.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_firmware.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_min_heap.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_user_copy.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_static_keys.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_static_key_base.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_printf.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_scanf.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_bitmap.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_uuid.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_xarray.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_maple_tree.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_memcat_p.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_meminit.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_ref_tracker.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_objpool.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/atomic64_test.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/virtio/virtio_dma_buf.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/ali-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/amd64-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/intel-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/intel-gtt.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/nvidia-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/sis-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/sworks-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/ttyprintk.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/lp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/panel/panel-orisetech-ota5601a.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/tiny/cirrus.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/drm_panel_orientation_quirks.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/drm_mipi_dbi.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/base/regmap/regmap-spi-avmm.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mfd/arizona.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mfd/pcf50633-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/cxl/cxl_pci.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/media/dvb-frontends/mb86a16.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mmc/core/sdio_uart.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-spilib.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-light.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-raw.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-audio-manager.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-gbphy.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-i2c.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-pwm.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-sdio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/devfreq/governor_simpleondemand.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/hwtracing/intel_th/intel_th_msu_sink.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/siox/siox-bus-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/vfio-mdev/mdpy-fb.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/vfio-mdev/mbochs.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/kobject/kobject-example.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/kobject/kset-example.o >> ERROR: modpost: "snd_pcm_set_sync_per_card" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 5/6] sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length
by kernel test robot 05 Jul '24

05 Jul '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 5f9f982dd71b418aeba7a0b37f87312545c06df4 commit: d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 [5/6] ALSA: pcm: optimize and clarify stream synchronization ID API config: x86_64-defconfig (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-lkp@…) compiler: gcc-13 (Ubuntu 13.2.0-4ubuntu3) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202407050501.o5Z3bibi-lkp@intel.com/ All warnings (new ones prefixed by >>): In function 'snd_pcm_set_sync_per_card', inlined from 'snd_pcm_lib_ioctl_sync_id' at sound/core/pcm_lib.c:1837:3, inlined from 'snd_pcm_lib_ioctl' at sound/core/pcm_lib.c:1863:10: >> sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length [-Wstringop-truncation] 547 | strncpy(params->sync + 4, id, len); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/strncpy +547 sound/core/pcm_lib.c 517 518 /** 519 * snd_pcm_set_sync_per_card - set the PCM sync id with card number 520 * @substream: the pcm substream 521 * @params: modified hardware parameters 522 * @id: identifier (max 12 bytes) 523 * @len: identifier length (max 12 bytes) 524 * 525 * Sets the PCM sync identifier for the card with zero padding. 526 * 527 * User space or any user should use this 16-byte identifier for a comparison only 528 * to check if two IDs are similar or different. Special case is the identifier 529 * containing only zeros. Interpretation for this combination is - empty (not set). 530 * The contents of the identifier should not be interpreted in any other way. 531 * 532 * The synchronization ID must be unique per clock source (usually one sound card, 533 * but multiple soundcard may use one PCM word clock source which means that they 534 * are fully synchronized). 535 * 536 * This routine composes this ID using card number in first four bytes and 537 * 12-byte additional ID. When other ID composition is used (e.g. for multiple 538 * sound cards), make sure that the composition does not clash with this 539 * composition scheme. 540 */ 541 void snd_pcm_set_sync_per_card(struct snd_pcm_substream *substream, 542 struct snd_pcm_hw_params *params, 543 const unsigned char *id, unsigned int len) 544 { 545 *(__u32 *)params->sync = cpu_to_le32(substream->pcm->card->number); 546 len = max(12, len); > 547 strncpy(params->sync + 4, id, len); 548 memset(params->sync + 4 + len, 0, 12 - len); 549 } 550 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 0/6] ASoC: codecs: lpass-rx-macro: Few code cleanups
by Krzysztof Kozlowski 04 Jul '24

04 Jul '24

Hi, Changes in v3: - New patch #1 to fix clang jump warning ("ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm") - Link to v2: https://lore.kernel.org/r/20240628-b4-qcom-audio-lpass-codec-cleanups-v2-0-… Changes in v2: - Use cleanup.h instead of devm(), therefore not adding Dmitry's review. - New patch #5. - Link to v1: https://lore.kernel.org/r/20240627-b4-qcom-audio-lpass-codec-cleanups-v1-0-… Improve a bit the Qualcomm LPASS RX macro driver and align similar parts of code with LPASS WSA macro driver for consistency. No external dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm ASoC: codecs: lpass-rx-macro: Simplify with cleanup.h ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgets ASoC: codecs: lpass-wsa-macro: Simplify with cleanup.h include/sound/soc-dapm.h | 2 +- sound/soc/codecs/lpass-macro-common.h | 5 +++ sound/soc/codecs/lpass-rx-macro.c | 63 +++++++++++++++++------------------ sound/soc/codecs/lpass-wsa-macro.c | 22 ++++++------ sound/soc/soc-dapm.c | 2 +- 5 files changed, 47 insertions(+), 47 deletions(-) --- base-commit: feca1ff0cd5ab7bc3990ec5a387d81d4dff88068 change-id: 20240627-b4-qcom-audio-lpass-codec-cleanups-27175f1d069f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 7

[A52] Pipeware errors
by GitHub issues - opened 04 Jul '24

04 Jul '24

alsa-project/alsa-plugins issue #59 was opened from ManuLinares: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4085 I'm getting this errors: ``` jul 03 21:12:38 thought pipewire[63110]: ALSA lib confmisc.c:165:(snd_config_get_card) Cannot get card index for %f jul 03 21:12:38 thought pipewire[63110]: spa.alsa: 'plug:{SLAVE="a52:0,'hw:%f,3'"}': playback open failed: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: 0x5b856abf1c30: can't get format: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: usage: node.name=<string> jul 03 21:14:21 thought wireplumber[63111]: wplua: [string "alsa.lua"]:182: attempt to concatenate a nil value (local 'node_name') stack traceback: [string "alsa.lua"]:182: in function <[string "alsa.lua"]:175> ``` could this be related to the this commit? https://github.com/alsa-project/alsa-plugins/commit/dfc65f2d1f239256c201c68… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/59 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v1] ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563
by Shenghao Ding 04 Jul '24

04 Jul '24

Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls, and keep the digital gain and analog gain in tas2781_snd_controls. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Rename driver_name from tas2781-codec to tasdev-codec, request from some tas2563 users since the driver supports both tas2563 and tas2781, the drvier name seemed unfair to tas2563. - Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls. --- sound/soc/codecs/tas2781-i2c.c | 29 ++++++++++++++++++++++------- 1 file changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e0e630f66101..b412f8f2e5d5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -104,7 +104,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_putvol(tas_priv, ucontrol, mc); } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -119,7 +119,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -140,6 +140,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -147,8 +152,6 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -590,6 +593,18 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); @@ -617,8 +632,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -759,7 +774,7 @@ MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); static struct i2c_driver tasdevice_i2c_driver = { .driver = { - .name = "tas2781-codec", + .name = "tasdev-codec", .of_match_table = of_match_ptr(tasdevice_of_match), #ifdef CONFIG_ACPI .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda-pcm: Limit the maximum number of periods by MAX_BDL_ENTRIES
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> The HDaudio specification Section 3.6.2 limits the number of BDL entries to 256. Make sure we don't allow more periods than this normative value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 9fb8521b896b..f6e24edd7adb 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -258,6 +258,12 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); + /* Limit the maximum number of periods to not exceed the BDL entries count */ + if (runtime->hw.periods_max > HDA_DSP_MAX_BDL_ENTRIES) + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, + runtime->hw.periods_min, + HDA_DSP_MAX_BDL_ENTRIES); + /* Only S16 and S32 supported by HDA hardware when used without DSP */ if (sdev->dspless_mode_selected) snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, -- 2.43.0

2 1

[PATCH] ASoC: SOF: Intel: hda: fix null deref on system suspend entry
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> When system enters suspend with an active stream, SOF core calls hw_params_upon_resume(). On Intel platforms with HDA DMA used to manage the link DMA, this leads to call chain of hda_dsp_set_hw_params_upon_resume() -> hda_dsp_dais_suspend() -> hda_dai_suspend() -> hda_ipc4_post_trigger() A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first, which clears hext_stream->link_substream, and then hda_ipc4_post_trigger() is called with a NULL snd_pcm_substream pointer. Fixes: 2b009fa0823c ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4") Link: https://github.com/thesofproject/linux/issues/5080 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c61d298ea6b3..1c823f9eea57 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -617,12 +617,6 @@ static int hda_dai_suspend(struct hdac_bus *bus) sdai = swidget->private; ops = sdai->platform_private; - ret = hda_link_dma_cleanup(hext_stream->link_substream, - hext_stream, - cpu_dai); - if (ret < 0) - return ret; - /* for consistency with TRIGGER_SUSPEND */ if (ops->post_trigger) { ret = ops->post_trigger(sdev, cpu_dai, @@ -631,6 +625,12 @@ static int hda_dai_suspend(struct hdac_bus *bus) if (ret < 0) return ret; } + + ret = hda_link_dma_cleanup(hext_stream->link_substream, + hext_stream, + cpu_dai); + if (ret < 0) + return ret; } } -- 2.43.0

2 1

[PATCH 0/1] Swapped l/r channels on ThinkBook 13x Gen4
by MadEnvel 04 Jul '24

04 Jul '24

Earlier commits introduced audio support for Lenovo ThinBook 13x Gen 4 that needed a patch due to missing _DSD config. Unfortunately, it looks like the left and right channels in the config are swapped. This patch fixes channel assignment. Tested on Lenovo ThinkBook 13x Gen4. MadEnvel (1): Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- 2.45.2

1 0

[PATCH] ASoC: SOF: ipc4-topology: Use single token list for the copiers
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> There is no need to keep separate token list for dai and 'common' copier token list when the 'common' list is actually the aif list, the SOF_COPIER_DEEP_BUFFER_TOKENS are not applicable for buffers. We could have separate lists for all types but it is probably simpler to just use a single list for all types of copiers. Function specific tokens will be only parsed by function specific code anyways. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 7755c01735f5..90f6856ee80c 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -3307,14 +3307,17 @@ static int sof_ipc4_link_setup(struct snd_sof_dev *sdev, struct snd_soc_dai_link return 0; } -static enum sof_tokens common_copier_token_list[] = { +/* Tokens needed for different copier variants (aif, dai and buffer) */ +static enum sof_tokens copier_token_list[] = { SOF_COMP_TOKENS, + SOF_COPIER_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, SOF_IN_AUDIO_FORMAT_TOKENS, SOF_OUT_AUDIO_FORMAT_TOKENS, - SOF_COPIER_DEEP_BUFFER_TOKENS, - SOF_COPIER_TOKENS, SOF_COMP_EXT_TOKENS, + + SOF_COPIER_DEEP_BUFFER_TOKENS, /* for AIF copier */ + SOF_DAI_TOKENS, /* for DAI copier */ }; static enum sof_tokens pipeline_token_list[] = { @@ -3322,16 +3325,6 @@ static enum sof_tokens pipeline_token_list[] = { SOF_PIPELINE_TOKENS, }; -static enum sof_tokens dai_token_list[] = { - SOF_COMP_TOKENS, - SOF_AUDIO_FMT_NUM_TOKENS, - SOF_IN_AUDIO_FORMAT_TOKENS, - SOF_OUT_AUDIO_FORMAT_TOKENS, - SOF_COPIER_TOKENS, - SOF_DAI_TOKENS, - SOF_COMP_EXT_TOKENS, -}; - static enum sof_tokens pga_token_list[] = { SOF_COMP_TOKENS, SOF_GAIN_TOKENS, @@ -3368,23 +3361,23 @@ static enum sof_tokens process_token_list[] = { static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_aif_out] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_in] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_out] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_buffer] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, -- 2.43.0

2 1

[PATCH v1] ALSA: hda: cs35l41: Fix missing Speaker ID GPIO description in _DSD
by Stefan Binding 04 Jul '24

04 Jul '24

Laptop 10431A63 contains valid _DSD, but missing Speaker ID description. Add this discription, but keep the rest of the _DSD to ensure the correct firmware and tuning is loaded for this laptop. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 86 ++++++++++++++++------------ sound/pci/hda/cs35l41_hda.h | 1 + sound/pci/hda/cs35l41_hda_property.c | 15 +++++ 3 files changed, 65 insertions(+), 37 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index ee9f83b737de..4b411ed8c3fe 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1753,38 +1753,14 @@ int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int return speaker_id; } -static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) +int cs35l41_hda_parse_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; u32 values[HDA_MAX_COMPONENTS]; - struct acpi_device *adev; - struct device *physdev; - struct spi_device *spi; - const char *sub; char *property; size_t nval; int i, ret; - adev = acpi_dev_get_first_match_dev(hid, NULL, -1); - if (!adev) { - dev_err(cs35l41->dev, "Failed to find an ACPI device for %s\n", hid); - return -ENODEV; - } - - cs35l41->dacpi = adev; - physdev = get_device(acpi_get_first_physical_node(adev)); - - sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); - if (IS_ERR(sub)) - sub = NULL; - cs35l41->acpi_subsystem_id = sub; - - ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); - if (!ret) { - dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); - goto out; - } - property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); if (ret <= 0) @@ -1816,8 +1792,9 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i /* To use the same release code for all laptop variants we can't use devm_ version of * gpiod_get here, as CLSA010* don't have a fully functional bios with an _DSD node */ - cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(adev), "reset", cs35l41->index, - GPIOD_OUT_LOW, "cs35l41-reset"); + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(cs35l41->dacpi), "reset", + cs35l41->index, GPIOD_OUT_LOW, + "cs35l41-reset"); property = "cirrus,speaker-position"; ret = device_property_read_u32_array(physdev, property, values, nval); @@ -1873,6 +1850,51 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i hw_cfg->bst_type = CS35L41_EXT_BOOST; hw_cfg->valid = true; + + return 0; +err: + dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); + hw_cfg->valid = false; + hw_cfg->gpio1.valid = false; + hw_cfg->gpio2.valid = false; + acpi_dev_put(cs35l41->dacpi); + + return ret; +} + +static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) +{ + struct acpi_device *adev; + struct device *physdev; + struct spi_device *spi; + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(cs35l41->dev, "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + + cs35l41->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + cs35l41->acpi_subsystem_id = sub; + + ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); + if (!ret) { + dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); + goto out; + } + + ret = cs35l41_hda_parse_acpi(cs35l41, physdev, id); + if (ret) { + put_device(physdev); + return ret; + } out: put_device(physdev); @@ -1888,16 +1910,6 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i } return 0; - -err: - dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); - hw_cfg->valid = false; - hw_cfg->gpio1.valid = false; - hw_cfg->gpio2.valid = false; - acpi_dev_put(cs35l41->dacpi); - put_device(physdev); - - return ret; } int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index b0bebb778462..c730b3351589 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -104,5 +104,6 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i struct regmap *regmap, enum control_bus control_bus); void cs35l41_hda_remove(struct device *dev); int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); +int cs35l41_hda_parse_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id); #endif /*__CS35L41_HDA_H__*/ diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 51998d1c72ff..5860379a0412 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -428,6 +428,20 @@ static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *phy return 0; } +static int missing_speaker_id_gpio2(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + int ret; + + ret = cs35l41_add_gpios(cs35l41, physdev, -1, 2, -1, 2); + if (ret) { + dev_err(cs35l41->dev, "Error adding GPIO mapping: %d\n", ret); + return ret; + } + + return cs35l41_hda_parse_acpi(cs35l41, physdev, id); +} + struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -501,6 +515,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "104317F3", generic_dsd_config }, { "CSC3551", "10431863", generic_dsd_config }, { "CSC3551", "104318D3", generic_dsd_config }, + { "CSC3551", "10431A63", missing_speaker_id_gpio2 }, { "CSC3551", "10431A83", generic_dsd_config }, { "CSC3551", "10431B93", generic_dsd_config }, { "CSC3551", "10431C9F", generic_dsd_config }, -- 2.43.0

2 1

Virtual birthday cards
by hobefaj342＠devncie.com 04 Jul '24

04 Jul '24

Birthday cards are a timeless way to express affection and celebrate the special day of a loved one. Whether it's a heartfelt message, a funny joke, or a beautiful design, birthday cards have the power to convey sentiments in a tangible and memorable way. In today's digital age, while traditional paper cards still hold their charm, online platforms like Sendwishonline.com offer a convenient and innovative way to send birthday greetings. Visit To https://sendwishonline.com/en/group-cards/happy-birthday-cards

1 0

[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

4 4

[PATCH 0/4] ASoC: tas*: Fix up GPIO usage
by Linus Walleij 04 Jul '24

04 Jul '24

The TI TAS drivers use some legacy GPIO code and headers, this series fixes it up. The TAS2781 is a special case since it adds a handful of lines of deviating code to reconfigure a GPIO line for IRQ mode and then never actually use the IRQ obtained in the code. Is the line used by autonomous hardware? I'm puzzled by this. Anyways the patch suggest how to solve this properly by fixing the parent irqchip and I'm happy to help. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (4): ASoC: tas5086: Convert to GPIO descriptors ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2552.c | 2 -- sound/soc/codecs/tas2764.c | 2 -- sound/soc/codecs/tas2770.c | 2 -- sound/soc/codecs/tas2780.c | 2 -- sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 26 ++++---------------------- sound/soc/codecs/tas5086.c | 27 ++++++++++++--------------- 10 files changed, 18 insertions(+), 56 deletions(-) --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240701-asoc-tas-gpios-5c051d80d768 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 6

[PATCH v3 0/6] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support". To apply this on the ASoC tree, a merge of Takashi's for-next branch is required. v3: add configuration with rt722 on Link3 v2: changed order of patches to fix git bisect error reported by Intel kbuild bot https://lore.kernel.org/oe-kbuild-all/202406131144.L6gW0I47-lkp@intel.com/ Bard Liao (3): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt722 l3 adr Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 145 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 ++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 ++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 290 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

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USB-Audio: Add support for Solid State Labs SSL 2+
by GitHub pull_request - opened 04 Jul '24

04 Jul '24

alsa-project/alsa-ucm-conf pull request #429 was opened from kescherCode: This adds support for the [Solid State Labs SSL 2+](https://www.solidstatelogic.com/products/ssl2-plus). Note that this *is* a distinct device to the SSL 2, support for which was added in #377. The most notable difference is that this device has two additional outputs (physically four, but two of those are mirrored). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/429 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/429.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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