January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v9 0/4] Media Device Allocator API
by shuah＠kernel.org 24 Jan '19

24 Jan '19

From: Shuah Khan <shuah(a)kernel.org> Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - This patch series is tested on 4.20-rc6 - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 705 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

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[alsa-devel] Possible ALSA loopback bug?
by Timothy Pearson 23 Jan '19

23 Jan '19

I've been running a dedicated machine with ALSA loopback, feeding darkice on the capture side and (currently) using mplayer to provide audio on the playback side. I've seen sporadic hangs for a couple of years on this setup, but recently with the switch to mplayer I have additional information that points more to ALSA than the other applications. When the audio hangs, mplayer prints a continuous stream of: "Audio device got stuck!" The only way to "unstick" the audio stream is to terminate and restart mplayer. Restarting darkice does not fix the stuck stream. There is no pulseaudio on this sytem. What would be the next steps to debug this setup? It's shown up on two entirely different installs at this point (Ubuntu 14.04 and Debian 9). Thanks! -- Timothy Pearson Raptor Engineering +1 (415) 727-8645 (direct line) +1 (512) 690-0200 (switchboard) https://www.raptorengineering.com

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[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Update broxton machine driver
by Yong Zhi 23 Jan '19

23 Jan '19

Hello, This patchset is an update to the following v2 posted by Naveen last year: https://patchwork.kernel.org/patch/10335295/ https://patchwork.kernel.org/patch/10335293/ https://patchwork.kernel.org/patch/10335297/ 1/5 and 5/5 of original series were merged. The changes have been tested for both bxt_da7219_max98357a and glk_da7219_max98357a. Yong Zhi (3): ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver ASoC: Intel: glk: Add DAI links for Multi-Playback sound/soc/intel/boards/bxt_da7219_max98357a.c | 101 +++++++++++++++++++++++--- 1 file changed, 89 insertions(+), 12 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: proc: Avoid possible leaks of snd_info_entry objects
by Takashi Iwai 23 Jan '19

23 Jan '19

This patch changes the parent pointer assignment of snd_info_entry object to be always non-NULL. More specifically,check the parent argument in snd_info_create_module_entry() & co, and assign snd_proc_root if NULL is passed there. This assures that the proc object is always freed when the root is freed, so avoid possible memory leaks. For example, some error paths (e.g. snd_info_register() error at snd_minor_info_init()) may leave snd_info_entry object although the proc file itself is freed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index fe502bc5e6d2..2dfb6389c084 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -741,7 +741,11 @@ struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry *parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = snd_proc_root; + entry = snd_info_create_entry(name, parent); if (entry) entry->module = module; return entry; @@ -762,7 +766,11 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry * parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = card->proc_root; + entry = snd_info_create_entry(name, parent); if (entry) { entry->module = card->module; entry->card = card; -- 2.16.4

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[alsa-devel] [PATCH] ALSA: pcm: Simplify proc file destruction
by Takashi Iwai 23 Jan '19

23 Jan '19

The proc files are recursively freed by calling with the root snd_info_entry object, so we don't have to keep each object for releasing one by one. Move the release of the PCM stream proc root at the beginning, so that we can remove the redundant code and resource. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 11 --------- sound/core/pcm.c | 66 +++++++++---------------------------------------- sound/core/pcm_memory.c | 16 ++---------- 3 files changed, 13 insertions(+), 80 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a20d3a48df00..eae6d2b82d7a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -481,15 +481,6 @@ struct snd_pcm_substream { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; - struct snd_info_entry *proc_hw_params_entry; - struct snd_info_entry *proc_sw_params_entry; - struct snd_info_entry *proc_status_entry; - struct snd_info_entry *proc_prealloc_entry; - struct snd_info_entry *proc_prealloc_max_entry; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_info_entry *proc_xrun_injection_entry; -#endif #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; @@ -511,10 +502,8 @@ struct snd_pcm_str { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG unsigned int xrun_debug; /* 0 = disabled, 1 = verbose, 2 = stacktrace */ - struct snd_info_entry *proc_xrun_debug_entry; #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index ca1ea3cf9350..bca0bdf3e33c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -536,12 +536,9 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry = snd_info_create_card_entry(pcm->card, "info", pstr->proc_root); if (entry) { snd_info_set_text_ops(entry, pstr, snd_pcm_stream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_info_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(pcm->card, "xrun_debug", @@ -551,24 +548,15 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry->c.text.write = snd_pcm_xrun_debug_write; entry->mode |= 0200; entry->private_data = pstr; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_xrun_debug_entry = entry; #endif return 0; } static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(pstr->proc_xrun_debug_entry); - pstr->proc_xrun_debug_entry = NULL; -#endif - snd_info_free_entry(pstr->proc_info_entry); - pstr->proc_info_entry = NULL; snd_info_free_entry(pstr->proc_root); pstr->proc_root = NULL; return 0; @@ -597,45 +585,33 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_info_entry = entry; entry = snd_info_create_card_entry(card, "hw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_hw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_hw_params_entry = entry; entry = snd_info_create_card_entry(card, "sw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_sw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_sw_params_entry = entry; entry = snd_info_create_card_entry(card, "status", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_status_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_status_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(card, "xrun_injection", @@ -645,40 +621,18 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; entry->mode = S_IFREG | 0200; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_xrun_injection_entry = entry; #endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } -static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) -{ - snd_info_free_entry(substream->proc_info_entry); - substream->proc_info_entry = NULL; - snd_info_free_entry(substream->proc_hw_params_entry); - substream->proc_hw_params_entry = NULL; - snd_info_free_entry(substream->proc_sw_params_entry); - substream->proc_sw_params_entry = NULL; - snd_info_free_entry(substream->proc_status_entry); - substream->proc_status_entry = NULL; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(substream->proc_xrun_injection_entry); - substream->proc_xrun_injection_entry = NULL; -#endif - snd_info_free_entry(substream->proc_root); - substream->proc_root = NULL; - return 0; -} #else /* !CONFIG_SND_VERBOSE_PROCFS */ static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; } -static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; } #endif /* CONFIG_SND_VERBOSE_PROCFS */ static const struct attribute_group *pcm_dev_attr_groups[]; @@ -911,15 +865,17 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif + + /* free all proc files under the stream */ + snd_pcm_stream_proc_done(pstr); + substream = pstr->substream; while (substream) { substream_next = substream->next; snd_pcm_timer_done(substream); - snd_pcm_substream_proc_done(substream); kfree(substream); substream = substream_next; } - snd_pcm_stream_proc_done(pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4b5356a10315..9a98bc61461f 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -93,12 +93,6 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { snd_pcm_lib_preallocate_dma_free(substream); -#ifdef CONFIG_SND_VERBOSE_PROCFS - snd_info_free_entry(substream->proc_prealloc_max_entry); - substream->proc_prealloc_max_entry = NULL; - snd_info_free_entry(substream->proc_prealloc_entry); - substream->proc_prealloc_entry = NULL; -#endif return 0; } @@ -203,21 +197,15 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) entry->c.text.write = snd_pcm_lib_preallocate_proc_write; entry->mode |= 0200; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_entry = entry; if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc_max", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_max_proc_read; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_max_entry = entry; } #else /* !CONFIG_SND_VERBOSE_PROCFS */ -- 2.16.4

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[alsa-devel] [PATCH v5 0/8] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 23 Jan '19

23 Jan '19

Thankyou for reviewing v4 patchset, here is v5 addressing comments from v4. Major change is moving the driver out of MFD as there are no immediate plans to add support to other drivers on this codec. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support without much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is our integration branch incase someone want to try this out: https://git.linaro.org/landing-teams/working/qualcomm/kernel.git/log/?h=int… Thanks, Srini Changes since v4 (https://lkml.org/lkml/2018/9/17/21/) - Move the driver out of MFD as discussed with Lee - defer support to MBHC for more testing! Srinivas Kandagatla (8): ASoC: dt-bindings: update wcd9335 bindings. ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add slim support .../bindings/sound/qcom,wcd9335.txt | 10 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 576 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5275 +++++++++++++++++ sound/soc/codecs/wcd9335.h | 640 ++ sound/soc/qcom/apq8096.c | 71 +- 8 files changed, 6623 insertions(+), 6 deletions(-) create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c create mode 100644 sound/soc/codecs/wcd9335.h -- 2.20.1

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[alsa-devel] How ALSA handle PCM playing when the ALSA modules are removing
by Zhu, Yingjiang 23 Jan '19

23 Jan '19

Hi, Takashi I am doing an ALSA module test with insert module/remove module loop, when playing a wave during this loop, there will be kernel oops. Does ALSA need handle this case? Or just forbid wave playing during the module load/unload? If this need to be handled, module unload will wait the wave play ending, or just stop the wave play at the first of the load/unload? BR Yingjiang

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[alsa-devel] [PATCH] ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
by shumingf＠realtek.com 22 Jan '19

22 Jan '19

From: Shuming Fan <shumingf(a)realtek.com> AD/DA ASRC function control two ASRC clock sources separately. Whether AD/DA filter select which clock source, we enable AD/DA ASRC function for all cases. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index bd45d03..9d5acd2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1784,7 +1784,9 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"ADC STO1 ASRC", NULL, "DA ASRC"}, {"ADC STO1 ASRC", NULL, "CLKDET"}, + {"DAC STO1 ASRC", NULL, "AD ASRC"}, {"DAC STO1 ASRC", NULL, "DA ASRC"}, {"DAC STO1 ASRC", NULL, "CLKDET"}, -- 2.7.4

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[alsa-devel] Applied "ASoC: core: Fix multi-CODEC setups" to the asoc tree
by Mark Brown 22 Jan '19

22 Jan '19

The patch ASoC: core: Fix multi-CODEC setups has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From 3f6a125230d8bfcbfe0c06ff0b8eaccbc727acd7 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Tue, 22 Jan 2019 17:36:11 +0000 Subject: [PATCH] ASoC: core: Fix multi-CODEC setups Revert 10dff9b0d (ASoC: soc-core: use for_each_link_codecs() for dai_link codecs) for now as Sylwester Nawrocki reports that it causes oopses on at least Odroid boards. Reported-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c92b4aff57b..8a58fa86675a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -870,7 +870,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_component *component; struct snd_soc_dai **codec_dais; @@ -905,8 +905,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + /* we can use for_each_rtd_codec_dai() after this */ codec_dais = rtd->codec_dais; - for_each_link_codecs(dai_link, i, codecs) { + for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_info(card->dev, "ASoC: CODEC DAI %s not registered\n", -- 2.20.1

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[alsa-devel] ALC255 Fixup (help needed)
by Stephan Lachnit 22 Jan '19

22 Jan '19

I have a Xiaomi Mi Air 2018 Laptop with one combined 3.5mm jack at the left side of the keyboard, connected via an ALC255. While audio output with jack detection works fine, alsa doesn't get any mic input. Internal mic and speakers work fine. I tried a bit around with hdajackretask and found that Pin ID 0x19 is wired to the mic input. I uploaded pictures of my config to my stack overflow question: https://stackoverflow.com/questions/54277467/how-to-contribute-to-alsa-inte… I know there should be a patch written for sound/pci/hda/patch_realtek.c which should look somewhat like this: [ALC255_FIXUP_XIAOMI_AIR18] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x01a1913c }, /* combinded audio jack */ }, but I really have no idea what the third and fourth line means and just copied from the Dell fixup. Would be nice if someone could help me to write a proper patch (I have some basic C/C++ knowledge, but no idea about alsa internals). System Info: Debian Unstable libasound2 1.1.7-2 i5 8250U w/ Realtek ALC255 This is my first attempt to contribute to open source project, so I hope I didn't miss anything important. Best regards, Stephan Lachnit

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