July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP machines
by Jeremy Szu 13 Jul '22

13 Jul '22

* The HP ProBook 440/450 G9 and EliteBook 640/650 G9 have multiple motherboard design and they are using different subsystem ID of audio codec. Add the same quirk for other MBs. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 007dd8b5e1f2..483c5f49af5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9096,6 +9096,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.25.1

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[PATCH v2 1/4] ALSA: hda: cs35l41: Don't dereference fwnode handle
by Andy Shevchenko 13 Jul '22

13 Jul '22

Use acpi_fwnode_handle() instead of dereferencing an fwnode handle directly, which is a better coding practice. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- v2: new change sound/pci/hda/cs35l41_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 49b25432a9f5..1a1afa0725e0 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -347,7 +347,7 @@ 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(&adev->fwnode, "reset", cs35l41->index, + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(adev), "reset", cs35l41->index, GPIOD_OUT_LOW, "cs35l41-reset"); property = "cirrus,speaker-position"; -- 2.35.1

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[PATCH 0/4] Extend ipc stream parameters sent to DSP
by Daniel Baluta 12 Jul '22

12 Jul '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need a way to send extra parameters to DSP firmware. In order to do this, we introduce ext_data array at the end of ipc_stream_params. With this new addition we can send compress parameters. This requires SOF ABI bump. Daniel Baluta (4): ASoC: SOF: compress: Dynamically allocate pcm params struct ASoC: SOF: Copy compress parameters into extended data ASoC: SOF: compress: Prevent current kernel running with older FW uapi: sof: abi: Bump SOF ABI for ext_data_length include/sound/sof/stream.h | 6 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/compress.c | 73 ++++++++++++++++++++++++------------ 3 files changed, 54 insertions(+), 27 deletions(-) -- 2.27.0

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[PATCH 0/2] ASoC: SOF: Intel: hda-dai: No need to decouple host/link DMA twice
by Peter Ujfalusi 12 Jul '22

12 Jul '22

Hi, There is no need to decouple a decoupled stream twice. Keep the decoupling in hda_link_stream_assign() only as it is going to be executed in all cases. Drop the outdated comment from hda_link_dma_hw_params() as well since the code has changed around it. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: Intel: hda-dai: Drop misleading comment regarding dma_data ASoC: SOF: Intel: hda-dai: Do snd_hdac_ext_stream_decouple() only once sound/soc/sof/intel/hda-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) -- 2.37.0

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[PATCH] ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used
by Peter Ujfalusi 12 Jul '22

12 Jul '22

The current implementation of probes only supports IPC3 and should not be loaded for other IPC implementation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-client-probes.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..e7e78d1a8ec3 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -694,6 +694,10 @@ static int sof_probes_client_probe(struct auxiliary_device *auxdev, if (!sof_probes_enabled) return -ENXIO; + /* only ipc3 is supported */ + if (sof_client_get_ipc_type(cdev) != SOF_IPC) + return -ENXIO; + if (!dev->platform_data) { dev_err(dev, "missing platform data\n"); return -ENODEV; -- 2.37.0

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[PATCH] ASoC: SOF: ipc3-topology: Prevent double freeing of ipc_control_data via load_bytes
by Peter Ujfalusi 12 Jul '22

12 Jul '22

We have sanity checks for byte controls and if any of the fail the locally allocated scontrol->ipc_control_data is freed up, but not set to NULL. On a rollback path of the error the higher level code will also try to free the scontrol->ipc_control_data which will eventually going to lead to memory corruption as double freeing memory is not a good thing. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 9448d5338423..b2cc046b9f60 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1644,6 +1644,7 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ return 0; err: kfree(scontrol->ipc_control_data); + scontrol->ipc_control_data = NULL; return ret; } -- 2.37.0

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[PATCH 0/3] ASoC: SOF: Intel: Do not process IPC reply before firmware boot
by Peter Ujfalusi 12 Jul '22

12 Jul '22

Hi, By mistake a developer managed to create a 'corrupted' IPC4 firmware image which loaded fine to the DSP and after boot it sent an IPC reply before we would have received the FW_READY message. It turned out that the image was an IPC3 firmware and the IPC reply was the IPC3 FW_READY notification message which got understood as an IPC4 reply message due to the difference between the two IPC mechanism. This caused a NULL pointer dereference since the reply memory will be allocated after the FW_READY message. To make sure this will not bite again, skip any spurious reply messages before the FW_READY. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot ASoC: SOF: Intel: mtl: Do not process IPC reply before firmware boot sound/soc/sof/intel/cnl.c | 37 +++++++++++++++++++++------------ sound/soc/sof/intel/hda-ipc.c | 39 ++++++++++++++++++++++------------- sound/soc/sof/intel/mtl.c | 20 +++++++++++------- 3 files changed, 62 insertions(+), 34 deletions(-) -- 2.37.0

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alsalib assertion using plug+dmix
by GitHub issues - opened 12 Jul '22

12 Jul '22

alsa-project/alsa-plugins issue #46 was opened from npitre: The espeak program is crashing randomly here. I've been able to isolate the issue in a small test program demonstrating the crash 100% of the time, ending in: ``` alsa_quickcrash: pcm.c:3236: snd_pcm_area_copy: Assertion `src < dst || src >= dst + bytes' failed. Aborted (core dumped) ``` Here's the test code: ``` /* to compile: gcc -O2 -Wall -o alsa_quickcrash alsa_quickcrash.c -lasound */ #include <alsa/asoundlib.h> #include <stdio.h> #include <string.h> #define DEVICE "test_dummy" #define RATE 22050 #define CHANNELS 1 #define FORMAT SND_PCM_FORMAT_S16_LE #define BUF_MS 60 #define DO(x) if (x < 0) { fprintf(stderr, "%s failed\n", #x); exit(1); } int main() { unsigned int rate = RATE; snd_pcm_uframes_t frames = RATE * BUF_MS / 1000; snd_pcm_t *handle; snd_pcm_hw_params_t *params; DO(snd_pcm_hw_params_malloc(&params)); DO(snd_pcm_open(&handle, DEVICE, SND_PCM_STREAM_PLAYBACK, 0)); DO(snd_pcm_hw_params_any(handle, params)); DO(snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED)); DO(snd_pcm_hw_params_set_format(handle, params, FORMAT)); DO(snd_pcm_hw_params_set_rate_near(handle, params, &rate, 0)); DO(snd_pcm_hw_params_set_channels(handle, params, CHANNELS)); DO(snd_pcm_hw_params_set_buffer_size_near(handle, params, &frames)); DO(snd_pcm_hw_params(handle, params)); DO(snd_pcm_prepare(handle)); printf("PCM config dump:\n"); snd_output_t *output = NULL; DO(snd_output_stdio_attach(&output, stdout, 0)); DO(snd_pcm_dump(handle, output)); printf("\n"); printf("frames = %ld\n", (long)frames); short data[frames * CHANNELS]; memset(data, 0, sizeof(data)); printf("writing whole buffer\n"); DO(snd_pcm_writei(handle, data, frames)); printf("writing whole buffer\n"); DO(snd_pcm_writei(handle, data, frames)); printf("writing whole buffer minus 1 frame\n"); DO(snd_pcm_writei(handle, data, frames - 1)); printf("draining audio\n"); DO(snd_pcm_drain(handle)); printf("done\n"); DO(snd_pcm_close(handle)); return 0; } ``` Here's the corresponding .asoundrc content: ``` pcm.test_dummy { type plug slave.pcm "mix_dummy" } pcm.mix_dummy { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0666 slave { pcm "hw:0" period_size 1024 buffer_size 4096 format "S16_LE" rate 48000 } } ``` And here's the complete test output: ``` PCM config dump: Plug PCM: Rate conversion PCM (48000, sformat=S16_LE) Converter: linear-interpolation Protocol version: 10003 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 22050 exact rate : 22050 (22050/1) msbits : 16 buffer_size : 1411 period_size : 470 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 470 period_event : 0 start_threshold : 1 stop_threshold : 1411 silence_threshold: 0 silence_size : 0 boundary : 3177289537109884928 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 1024 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 2 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 3072 period_size : 1024 period_time : 21333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 2 stop_threshold : 3072 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 1 'HDA Intel PCH' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 4096 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 0 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 2734916609 frames = 1411 writing whole buffer writing whole buffer writing whole buffer minus 1 frame draining audio alsa_quickcrash: pcm.c:3236: snd_pcm_area_copy: Assertion `src < dst || src >= dst + bytes' failed. Aborted (core dumped) ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/46 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH] ASoC: soc-pcm: demote warnings on non-atomic BE connection
by Pierre-Louis Bossart 12 Jul '22

12 Jul '22

When an FE, typically non-atomic, is connected to an atomic BE, we force the BE as non-atomic. There's no reason to throw a warning, this is a perfectly fine configuration and a conversion that's required by-design. This removes the unconditional warnings such as [ 12.054213] iDisp1: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.074693] iDisp2: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.096612] iDisp3: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.118637] iDisp4: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.140660] dmic01: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic [ 12.147521] dmic16k: dpcm_be_connect: FE is nonatomic but BE is not, forcing BE as nonatomic and demotes them to dev_dbg(), as suggested in review comments. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..5b99bf2dbd08 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1209,8 +1209,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, return -EINVAL; } if (fe_substream->pcm->nonatomic && !be_substream->pcm->nonatomic) { - dev_warn(be->dev, "%s: FE is nonatomic but BE is not, forcing BE as nonatomic\n", - __func__); + dev_dbg(be->dev, "FE is nonatomic but BE is not, forcing BE as nonatomic\n"); be_substream->pcm->nonatomic = 1; } -- 2.34.1

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[PATCH] ASoC: SOF: topology: remove unused variable
by Peter Ujfalusi 12 Jul '22

12 Jul '22

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> 'ret' is never used. Remove it and return 0 instead. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 7e54eb1bf77b..9273a70fec25 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1419,7 +1419,6 @@ static int sof_widget_unload(struct snd_soc_component *scomp, struct soc_bytes_ext *sbe; struct snd_sof_dai *dai; struct soc_enum *se; - int ret = 0; int i; swidget = dobj->private; @@ -1480,7 +1479,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, list_del(&swidget->list); kfree(swidget); - return ret; + return 0; } /* -- 2.37.0

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