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

[PATCH] ALSA: hda/realtek: Fix deadlock by COEF mutex
by Takashi Iwai 14 Feb '22

14 Feb '22

The recently introduced coef_mutex for Realtek codec seems causing a deadlock when the relevant code is invoked from the power-off state; then the HD-audio core tries to power-up internally, and this kicks off the codec runtime PM code that tries to take the same coef_mutex. In order to avoid the deadlock, do the temporary power up/down around the coef_mutex acquisition and release. This assures that the power-up sequence runs before the mutex, hence no re-entrance will happen. Fixes: b837a9f5ab3b ("ALSA: hda: realtek: Fix race at concurrent COEF updates") Reported-and-tested-by: Julian Wollrath <jwollrath(a)web.de> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20220214132838.4db10fca@schienar Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 39 +++++++++++++++++++++-------------- 1 file changed, 24 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9473fb76ff19..3a42457984e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -138,6 +138,22 @@ struct alc_spec { * COEF access helper functions */ +static void coef_mutex_lock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_power_up_pm(codec); + mutex_lock(&spec->coef_mutex); +} + +static void coef_mutex_unlock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + mutex_unlock(&spec->coef_mutex); + snd_hda_power_down_pm(codec); +} + static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { @@ -151,12 +167,11 @@ static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { - struct alc_spec *spec = codec->spec; unsigned int val; - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); val = __alc_read_coefex_idx(codec, nid, coef_idx); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); return val; } @@ -173,11 +188,9 @@ static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int coef_val) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_write_coefex_idx(codec, nid, coef_idx, coef_val); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_write_coef_idx(codec, coef_idx, coef_val) \ @@ -198,11 +211,9 @@ static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int mask, unsigned int bits_set) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_update_coef_idx(codec, coef_idx, mask, bits_set) \ @@ -235,9 +246,7 @@ struct coef_fw { static void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); for (; fw->nid; fw++) { if (fw->mask == (unsigned short)-1) __alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); @@ -245,7 +254,7 @@ static void alc_process_coef_fw(struct hda_codec *codec, __alc_update_coefex_idx(codec, fw->nid, fw->idx, fw->mask, fw->val); } - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } /* -- 2.31.1

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[PATCH] ALSA: usb-audio: Don't abort resume upon errors
by Takashi Iwai 14 Feb '22

14 Feb '22

The default mixer resume code treats the errors at restoring the modified mixer items as a fatal error, and it returns back to the caller. This ends up in the resume failure, and the device will be come unavailable, although basically those errors are intermittent and can be safely ignored. The problem itself has been present from the beginning, but it didn't hit usually because the code tries to resume only the modified items. But now with the recent commit to forcibly initialize each item at the probe time, the problem surfaced more often, hence it appears as a regression. This patch fixes the regression simply by ignoring the errors at resume. Fixes: b96681bd5827 ("ALSA: usb-audio: Initialize every feature unit once at probe time") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215561 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 630766ba259f..a5641956ef10 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3678,17 +3678,14 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) - return err; + break; } idx++; } } else { /* master */ - if (cval->cached) { - err = snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); - if (err < 0) - return err; - } + if (cval->cached) + snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); } return 0; -- 2.31.1

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[PATCH] ASoC: SOF: compr: Mark snd_compress_ops static
by Daniel Baluta 14 Feb '22

14 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Functions won't be directly used outside of compress.c file so mark them as static. This will also fix warnings reported by kernel test robot: >> sound/soc/sof/compress.c:91:5: warning: no previous prototype for function 'sof_compr_open' [-Wmissing-prototypes] int sof_compr_open(struct snd_soc_component *component, ^ sound/soc/sof/compress.c:91:1: note: declare 'static' if the function is not intended to be used outside of this translation unit int sof_compr_open(struct snd_soc_component *component, Fixes: 6324cf901e14 ("SoC: SOF: compr: Add compress ops implementation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 2137d1d65dd9..2af8d75204e9 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -89,8 +89,8 @@ static int create_page_table(struct snd_soc_component *component, spcm->stream[dir].page_table.area, size); } -int sof_compr_open(struct snd_soc_component *component, - struct snd_compr_stream *cstream) +static int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *crtd = cstream->runtime; @@ -125,8 +125,8 @@ int sof_compr_open(struct snd_soc_component *component, return 0; } -int sof_compr_free(struct snd_soc_component *component, - struct snd_compr_stream *cstream) +static int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; @@ -159,8 +159,8 @@ int sof_compr_free(struct snd_soc_component *component, return ret; } -int sof_compr_set_params(struct snd_soc_component *component, - struct snd_compr_stream *cstream, struct snd_compr_params *params) +static int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -222,8 +222,8 @@ int sof_compr_set_params(struct snd_soc_component *component, return 0; } -int sof_compr_get_params(struct snd_soc_component *component, - struct snd_compr_stream *cstream, struct snd_codec *params) +static int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) { /* TODO: we don't query the supported codecs for now, if the * application asks for an unsupported codec the set_params() will fail. @@ -231,8 +231,8 @@ int sof_compr_get_params(struct snd_soc_component *component, return 0; } -int sof_compr_trigger(struct snd_soc_component *component, - struct snd_compr_stream *cstream, int cmd) +static int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -271,9 +271,9 @@ int sof_compr_trigger(struct snd_soc_component *component, &reply, sizeof(reply)); } -int sof_compr_copy(struct snd_soc_component *component, - struct snd_compr_stream *cstream, - char __user *buf, size_t count) +static int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) { struct snd_compr_runtime *rtd = cstream->runtime; unsigned int offset, n; -- 2.27.0

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[PATCH 1/3] ASoC: wm_adsp: Correct control read size when parsing compressed buffer
by Charles Keepax 14 Feb '22

14 Feb '22

When parsing the compressed stream the whole buffer descriptor is now read in a single cs_dsp_coeff_read_ctrl; on older firmwares this descriptor is just 4 bytes but on more modern firmwares it is 24 bytes. The current code reads the full 24 bytes regardless, this was working but reading junk for the last 20 bytes. However commit f444da38ac92 ("firmware: cs_dsp: Add offset to cs_dsp read/write") added a size check into cs_dsp_coeff_read_ctrl, causing the older firmwares to now return an error. Update the code to only read the amount of data appropriate for the firmware loaded. Fixes: 04ae08596737 ("ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3672e3d1703e..0582585236a28 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1441,7 +1441,8 @@ static int wm_adsp_buffer_parse_coeff(struct cs_dsp_coeff_ctl *cs_ctl) int ret, i; for (i = 0; i < 5; ++i) { - ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, sizeof(coeff_v1)); + ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, + min(cs_ctl->len, sizeof(coeff_v1))); if (ret < 0) return ret; -- 2.11.0

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[PATCH] ALSA: pcm: accept the OPEN state for snd_pcm_stop()
by Jaroslav Kysela 14 Feb '22

14 Feb '22

It may be useful to completely invalidate streaming under some circumstances like the USB gadget detach. This case is a bit different than the complete disconnection. The applications should be notified that the PCM streaming is no longer available, but the recovery may be expected. This patch adds support for SNDRV_PCM_STATE_OPEN passed to snd_pcm_stop() function. Only the hw_free ioctl is allowed to free the buffers in this state for a full recovery operation or the PCM file handle must be closed. In the OPEN state, ioctls return EBADFD error (with the added hw_free exception above). The applications which are not aware about this new state transition (and recovery) will fail with an error. This operation is expected. Link: https://lore.kernel.org/alsa-devel/4e17c99b-1d8a-8ebe-972c-9b1299952ff8@ivi… Cc: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_native.c | 12 +++++++++++- 3 files changed, 13 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755..4de1c2c91109 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -467,6 +467,7 @@ struct snd_pcm_substream { /* -- assigned files -- */ int ref_count; atomic_t mmap_count; + atomic_t queued_hw_free; unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6fd3677685d7..8dc7e99b2113 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -694,6 +694,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) snd_pcm_group_init(&substream->self_group); list_add_tail(&substream->link_list, &substream->self_group.substreams); atomic_set(&substream->mmap_count, 0); + atomic_set(&substream->queued_hw_free, 0); prev = substream; } return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..118ad3f26f4a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -686,10 +686,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: + if (atomic_read(&substream->queued_hw_free)) + goto __badfd; case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: break; default: +__badfd: snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -829,6 +832,7 @@ static int do_hw_free(struct snd_pcm_substream *substream) result = substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) snd_pcm_lib_free_pages(substream); + atomic_set(&substream->queued_hw_free, 0); return result; } @@ -1454,6 +1458,8 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); + if (state == SNDRV_PCM_STATE_OPEN) + atomic_set(&substream->queued_hw_free, 1); } static const struct action_ops snd_pcm_action_stop = { @@ -1469,6 +1475,9 @@ static const struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * + * If the requested state is OPEN, the stream is invalidated and + * the application must call hw_free to recover the operation. + * * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) @@ -2637,7 +2646,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN || + atomic_read(&substream->queued_hw_free)) do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.33.1

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[PATCH 1/2] ALSA: hda: Fix regression on forced probe mask option
by Takashi Iwai 14 Feb '22

14 Feb '22

The forced probe mask via probe_mask 0x100 bit doesn't work any longer as expected since the bus init code was moved and it's clearing the codec_mask value that was set beforehand. This patch fixes the long-time regression by moving the check_probe_mask() call. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Reported-by: dmummenschanz(a)web.de Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/trinity-f018660b-95c9-442b-a2a8-c92a56eb07ed-1644… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4b0338c4c543..18b795220b52 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1798,8 +1798,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, assign_position_fix(chip, check_position_fix(chip, position_fix[dev])); - check_probe_mask(chip, dev); - if (single_cmd < 0) /* allow fallback to single_cmd at errors */ chip->fallback_to_single_cmd = 1; else /* explicitly set to single_cmd or not */ @@ -1825,6 +1823,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->bus.core.needs_damn_long_delay = 1; } + check_probe_mask(chip, dev); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device [card]!\n"); -- 2.31.1

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Cannon Lake PCH cAVS (ALC892) not detected by kernel driver
by dmummenschanz＠web.de 14 Feb '22

14 Feb '22

Hello, the audio chip Cannon Lake PCH cAVS ALC892 does not get recognized on my system. Only the HDMI devices are recognized. lspci shows the following audio device: 00:1f.3 Audio device: Intel Corporation Cannon Lake PCH cAVS (rev 10) dmesg only shows: [ 189.559786] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_exit [i915]) [ 189.581395] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 189.581503] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 189.581628] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 [ 189.581711] input: HDA Intel PCH HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 [ 189.581849] input: HDA Intel PCH HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input24 ALSA-info file: http://alsa-project.org/db/?f=96102ee84f258d1cac83ed33795356550181511c There is a workaround availble: echo auto | tee /sys/devices/pci0000\:00/0000\:00\:1f.3/power/control echo 1 | tee /sys/devices/pci0000\:00/0000\:00\:1f.3/remove echo 1 | tee /sys/bus/pci/rescan unfortunately this doesn't work for me. The device never shows up. Sound on winX works fine on this device btw. I'm happy to assist with any attempts to make the sound work.

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[PATCH 0/4] ALSA: hda: Expose codec organization functions
by Cezary Rojewski 14 Feb '22

14 Feb '22

Changes expose several function that are currently unavailable for HDA-DSP drivers for use. Those functions are: snd_hda_codec_cleanup_for_unbind() snd_hda_codec_set_power_save() snd_hda_codec_register() snd_hda_codec_unregister() snd_hda_codec_device_init() This allows upcoming AVS driver [1] to re-use even mode of HDA related code that is currently available in sound/pci/hda and sound/hda and prevent any code duplication within avs-driver that would otherwise had to happen. Last patch in the series provides snd_hdac_ext_bus_link_at() - a helper function which allows for retrieval of HDA segment (link) based on codec address directly. This is simpler than parsing codec-name first to extract the address what is the case for snd_hdac_ext_bus_get_link(). The latter function is updated to re-use newly added one so core logic is not duplicated after the addition. [1]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (4): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions include/sound/hda_codec.h | 11 +++- include/sound/hdaudio_ext.h | 1 + sound/hda/ext/hdac_ext_controller.c | 31 +++++++--- sound/pci/hda/hda_codec.c | 94 ++++++++++++++++++++--------- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- 6 files changed, 100 insertions(+), 41 deletions(-) -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Legion Y9000X 2019
by Yu Huang 14 Feb '22

14 Feb '22

Legion Y9000X 2019 has the same speaker with Y9000X 2020, but with a different quirk address. Add one quirk entry to make the speaker work on Y9000X 2019 too. Signed-off-by: Yu Huang <diwang90(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 8315bf7d4c38..9473fb76ff19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9170,6 +9170,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), SND_PCI_QUIRK(0x17aa, 0x3847, "Legion 7 16ACHG6", ALC287_FIXUP_LEGION_16ACHG6), SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.34.1

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Re: [PATCH 1/1] ALSA: usb-audio: revert to IMPLICIT_FB_FIXED_DEV for M-Audio FastTrack Ultra
by Takashi Iwai 14 Feb '22

14 Feb '22

On Fri, 11 Feb 2022 23:49:13 +0100, Matteo Martelli wrote: > > Commit 83b7dcbc51c930fc2079ab6c6fc9d719768321f1 introduced a generic > implicit feedback parser, which fails to execute for M-Audio FastTrack > Ultra sound cards. The issue is with the ENDPOINT_SYNCTYPE check in > add_generic_implicit_fb() where the SYNCTYPE is ADAPTIVE instead of ASYNC. > The reason is that the sync type of the FastTrack output endpoints are > set to adaptive in the quirks table since commit > 65f04443c96dbda11b8fff21d6390e082846aa3c. > > Signed-off-by: Matteo Martelli <matteomartelli3(a)gmail.com> Thanks, applied now. Takashi

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