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[PATCH] ALSA: oxfw: fix transmission method for Loud models based on OXFW971
by Takashi Sakamoto 13 Sep '21

13 Sep '21

Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } -- 2.30.2

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[PATCH AUTOSEL 4.4 29/35] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 13 Sep '21

13 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index d0d338533eb6..64079423b298 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -134,7 +134,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

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[PATCH AUTOSEL 5.13 176/219] ASoC: intel: atom: Revert PCM buffer address setup workaround again
by Sasha Levin 11 Sep '21

11 Sep '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e28ac04a705e946eddc5e7d2fc712dea3f20fe9e ] We worked around the breakage of PCM buffer setup by the commit 65ca89c2b12c ("ASoC: intel: atom: Fix breakage for PCM buffer address setup"), but this isn't necessary since the CONTINUOUS buffer type also sets runtime->dma_addr since commit f84ba106a018 ("ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too"). Let's revert the change again. Link: https://lore.kernel.org/r/20210822072127.9786-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 905c7965f653..5db2f4865bbb 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); + u32 buffer_addr = substream->runtime->dma_addr; channels = substream->runtime->channels; period_size = substream->runtime->period_size; -- 2.30.2

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Raw ALSA PCM
by Ryan McClue 11 Sep '21

11 Sep '21

Hello, I'm trying to play a simple PCM buffer with ALSA without asoundlib. This has proven very difficult as there are very few resources that show usage without asoundlib. Through my investigations I was fortunate enough to obtain a gist that attempts to do this: https://gist.github.com/takaswie/d8a9553c230971d974a3dfc759b2ff67 Unfortunately this application does not work. I tested the application using the virtual snd-dummy driver and the application produced the same error. So, it is most likely an application error, rather than an Intel HDA error. I'm emailing to ask for help in getting the program to work. I'm really lost as to what should be changed in the code. I'm sure that a working example of this will be useful for many programmers. Thank you

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ASoC/ALSA Out of Memory Issue
by Alex Roberts 10 Sep '21

10 Sep '21

Hello, I am developing a dummy codec to interface with an 8-channel, 24-bit ADC. I've got it working on an NXP imx8m through the fsl_sai driver on kernel 5.4.85. I can capture all 8 channels at varying sample rates using arecord, and I've verified correct data capture via opening the resulting .wav file in Audacity. The problem I am having is that occasionally, upon starting arecord - after a fresh power cycle - I get an out of memory error. Other times I get an out of memory after a non-deterministic period of capture. Starting capture again also reports out of memory, but if I wait several minutes and start capture it will start recording again. A power cycle usually helps, but as stated earlier, not 100% of the time. I'm trying to track down where the oom error is coming from, but haven't had much luck. My colleague tried running arecord with valgrind to check for memory leaks and nothing of note was observed. My suspicion is there's something going on with allocated memory for DMA, like fragmentation starts to happen and it can't get a contiguous region for operation. Reserving a larger pool - either via device tree or kernel cmdline arguments in the bootoader - did not seem to help. Another thought is that it's a boundary/alignment issue due to the 24-bit data, and the error is the result of trying to allocate a chunk of memory for DMA that doesn't align. I'm very new to ALSA dev with some exposure to kernel dev in general, so please correct me if I'm wrong or completely mis-understanding something. Any suggestions on where I should / how I can debug this memory error? Thanks, Alex. PS: Previously sent this to just alsa-devel mailing list on 7/21, but never saw it show up in the archives. Here is more info since then: The goal is 8-channel, 96k sampling rate. I've reduced sampling rate and still have the issue. Reducing down to 4-channels helps, but haven't tested long term enough to evaluate by how much. Narrowed it down to device_prep_dma_cyclic(..) returning NULL within dmaengine_prep_dma_cyclic(..)..... still tracing through source to learn exactly what is going on.

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[PATCH] ASoC: mediatek: common: handle NULL case in suspend/resume function
by Trevor Wu 10 Sep '21

10 Sep '21

When memory allocation for afe->reg_back_up fails, reg_back_up can't be used. Keep the suspend/resume flow but skip register backup when afe->reg_back_up is NULL, in case illegal memory access happens. Fixes: 283b612429a2 ("ASoC: mediatek: implement mediatek common structure") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index baaa5881b1d4..e95c7c018e7d 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component) devm_kcalloc(dev, afe->reg_back_up_list_num, sizeof(unsigned int), GFP_KERNEL); - for (i = 0; i < afe->reg_back_up_list_num; i++) - regmap_read(regmap, afe->reg_back_up_list[i], - &afe->reg_back_up[i]); + if (afe->reg_back_up) { + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + } afe->suspended = true; afe->runtime_suspend(dev); @@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component) afe->runtime_resume(dev); - if (!afe->reg_back_up) + if (!afe->reg_back_up) { dev_dbg(dev, "%s no reg_backup\n", __func__); - - for (i = 0; i < afe->reg_back_up_list_num; i++) - mtk_regmap_write(regmap, afe->reg_back_up_list[i], - afe->reg_back_up[i]); + } else { + for (i = 0; i < afe->reg_back_up_list_num; i++) + mtk_regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + } afe->suspended = false; return 0; -- 2.18.0

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[PATCH for-5.15 0/5] ASoC: fsl: register platform component before registering cpu dai
by Shengjiu Wang 10 Sep '21

10 Sep '21

There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. This patch set is to fix this issue for SAI, ESAI, MICFIL, SPDIF, XCVR drivers. Shengjiu Wang (5): ASoC: fsl_sai: register platform component before registering cpu dai ASoC: fsl_esai: register platform component before registering cpu dai ASoC: fsl_micfil: register platform component before registering cpu dai ASoC: fsl_spdif: register platform component before registering cpu dai ASoC: fsl_xcvr: register platform component before registering cpu dai sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- 5 files changed, 48 insertions(+), 26 deletions(-) -- 2.17.1

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[PATCH] ALSA: usb-audio: Unify mixer resume and reset_resume procedure
by Takashi Iwai 10 Sep '21

10 Sep '21

USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu <enshuo(a)chromium.org> Cc: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsM… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd570a42f043..1764b9302d46 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1100,20 +1100,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 43bc59575a6e..a2ce535df14b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 876bbc9a71ad..98ea24d91d80 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a66ce0375fd9..46082dc57be0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); -- 2.26.2

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A portion of USB Headsets loses previous sound volume setting after a suspend resume
by En-Shuo Hsu 10 Sep '21

10 Sep '21

Hi We recently found that some USB headsets may fall back to their full volume after a suspend and resume. We think the issue is caused by the logic of mixer_ctl_feature_put <https://github.com/torvalds/linux/blob/a3fa7a101dcff93791d1b1bdb3affcad1410…> in sound/usb/mixer.c: err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &oval); if (err < 0) return filter_error(cval, err); val = ucontrol->value.integer.value[cnt]; val = get_abs_value(cval, val); if (oval != val) { snd_usb_set_cur_mix_value(cval, c + 1, cnt, val); changed = 1; } The existing codes get the existing mixer control value and ignore the set if the val doesn't change. However, in the suspend and resume case, the USB headset's control value is actually changed. Removing the cache logic is a potential fix, but a better solution may be to properly handle the suspend resume scenario in snd_usb_get_cur_mix_value. We may need to mark the cache in usb_mixer_elem_info to be dirty. The issue is verified to be reproduced with Dell WH3022 and Logitech USB Headset H340 by: 1. Boot to OS. 2. Plug in the headset and check sound output. 3. Play an audio/video and keep with low volume. 4. Suspend. 5. Resume. 6. When audio/video is played, the headset's sound output can't keep the original volume. --> issue " Would like to know your thoughts on this issue.

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[PATCH v3] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Randy Dunlap 10 Sep '21

10 Sep '21

On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Fixes: 3951e4aae2ce ("ASoC: atmel-pcm: dma support based on pcm dmaengine") Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Fixes: 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Cc: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- v2: rebase & resend; drop Cc: to Bo Shen <voice.shen(a)atmel.com> # bounced v3: drop "depends on HAS_DMA" (as suggested by Codrin; Thanks) --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) --- linux-next-20210707.orig/sound/soc/atmel/Kconfig +++ linux-next-20210707/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool

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