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

[PATCH AUTOSEL 6.11 13/16] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 63c1c87993e0e5bb11bced3d8224446a2bc62338 ] This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ad2492efb1cdc..19307812ec765 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; -- 2.43.0

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[PATCH AUTOSEL 6.11 07/16] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Sasha Levin 12 Nov '24

12 Nov '24

From: Shenghao Ding <shenghao-ding(a)ti.com> [ Upstream commit fe09de2db2365eed8b44b572cff7d421eaf1754a ] Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f07104..6474cc551d551 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.43.0

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[RESEND PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 12 Nov '24

12 Nov '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 | Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4082 - ASRock X870 Steel Legend
by GitHub issues - opened 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was opened from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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`amixer sset Master` doesnt increase and decrease both sides equally
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #279 was opened from lquenti: I look through the issues, man pages and internet and hope I didn't miss anything. Please tell me if its a user error. Whenever I use `amixer sset Master` to increase volume, it will not increase them evenly. Here is a before ![before](https://github.com/user-attachments/assets/4ccfa220-608c-4407-9f6e-fc1d27f1a91b) and after ![after](https://github.com/user-attachments/assets/250ab254-d46a-44e4-a201-58f9224b6293) Interestingly amixer seem to think that it is still evenly balanced, although alsamixer shows that it is unbalanced afterwards. --- Interestingly, when decreasing, the unevenness flips to the other side. [Other people also seem to struggle with it](https://stackoverflow.com/questions/69826139/using-amixer-to-set-audio-… Thank you so much. If it is something somewhat simple to fix, I can also write a fix myself, but for now I am unsure what the problem is. Issue URL : https://github.com/alsa-project/alsa-utils/issues/279 Repository URL: https://github.com/alsa-project/alsa-utils

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Change 26ce:0a08 to list multiple motherboards
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #461 was opened from cybik: The hardware id `26ce:0a08` seems to be re-used by multiple motherboards at ASRock. Identify them within `[brackets]` to point out the specific boards in question. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ThinkPad P14s Gen 5 AMD micmute LED is not working sometimes.
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #278 was opened from pma1: The mic and sound output works fine. The mute LED works fine. But when press Fn+F4, the mic-mute led is not working sometimes after bootup. The failrate is like 1/10. Tested with latest alsa-ucm from git, it's reproduced. /sys/class/sound/ctl-led/mic/card1# cat list 7 # amixer -c 1 controls numid=12,iface=CARD,name='Headphone Jack' numid=11,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Phantom Jack' numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=2,iface=MIXER,name='Headphone Playback Switch' numid=1,iface=MIXER,name='Headphone Playback Volume' numid=16,iface=MIXER,name='PCM Playback Volume' numid=17,iface=MIXER,name='Mic ACP LED Capture Switch' numid=8,iface=MIXER,name='Mic Boost Volume' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=5,iface=MIXER,name='Auto-Mute Mode' numid=4,iface=MIXER,name='Speaker Playback Switch' numid=3,iface=MIXER,name='Speaker Playback Volume' numid=15,iface=PCM,name='Capture Channel Map' numid=14,iface=PCM,name='Playback Channel Map' # echo 'Capture Switch' > detach # echo 'Mic ACP LED Capture Switch' > attach /sys/class/sound/ctl-led/mic/card1# cat list 17 Then the micmute LED will work. [alsa-info-amd-micmutled.txt](https://github.com/user-attachments/files/1627… Issue URL : https://github.com/alsa-project/alsa-utils/issues/278 Repository URL: https://github.com/alsa-project/alsa-utils

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USB-Audio: ALC4080: Add support for MSI MPG X870E CARBON WIFI (0db0:0b58)
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #460 was opened from xatier: This PR updates the regex for MSI MPG X870E CARBON WIFI motherboard. ``` $ lsusb | ag audio Bus 003 Device 007: ID 0db0:0b58 Micro Star International USB Audio ``` ``` Audio Realtek® ALC4080 Codec ``` https://www.msi.com/Motherboard/MPG-X870E-CARBON-WIFI/Specification Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 6.1.y] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3
by Werner Sembach 09 Nov '24

09 Nov '24

From: Christoffer Sandberg <cs(a)tuxedo.de> Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 a8bc95ffa41a3..da14ab97783f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10001,6 +10001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.43.0

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Odd behavior with Tempotec Serenade PCIe (CM8888)
by GitHub issues - opened 09 Nov '24

09 Nov '24

alsa-project/alsa-lib issue #418 was opened from lss4: This card is powered by CM8888, using `snd-hda-intel` driver, and it does work in Linux, but its behavior is quite odd. 1. Both the RCA connection and 6.35 headphone jack on the card are connected at the moment. Preferably I'd like to be able to switch to use either RCA (to speakers) or headphone, but I can't -- Audio always goes through both outputs, and there's only one analog output from this card. 2. This is the odd behavior I'm talking about -- the card's outputs are muted by default. Playing an audio file will trigger the card to unmute the output after 2-3 seconds followed by an audible click from the relay on the sound card. When the output is active, if no audio is being actively played for at least 10 seconds, the output will be muted again on its own, and this is again followed by an audible click from the relay. Honestly I've never seen an audio card capable of doing something that "intelligent" before. `alsa-info.sh` output attached here. The sound card in question is `card1` (`hwC1D1`). [alsa-info-20241109.txt](https://github.com/user-attachments/files/17687665/… I'm currently using Garuda Linux on this system. Previously I was using Manjaro. Both are Arch-based, and the behaviors are the same. I'm not sure about the support of this chip (CM8888) in general, as the information I could find online about CM8888 on Linux is limited and mostly outdated (several years ago). Issue URL : https://github.com/alsa-project/alsa-lib/issues/418 Repository URL: https://github.com/alsa-project/alsa-lib

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[Question] About Roland Rubix series, alsa-ucm-conf and snd_usb_audio
by GitHub issues - opened 08 Nov '24

08 Nov '24

alsa-project/alsa-ucm-conf issue #459 was opened from sim590: I want to understand what `alsa_ucm-conf` is about exactly regarding the support for specific devices. I'm thinking to buy a Roland Rubix series audio interface (either 22 or 24) and I'm wondering if both my XLR inputs are gonna appear in alsa/pulseaudio/pipewire as separate input lines. Does that require a potential change in `snd_usb_audio` kernel module or only a configuration file on this repository? I've looked at the configuration files on this repository and I have found a Roland directory, but nothing about the Rubix series explicitly. Presently, I have a Focusrite Scarlett 2i2 *fake clone* (unfortunately) which I'm gonna return shortly because it's counterfeit. As of right now, I don't see the two separate outputs, but only the following input: ![image](https://github.com/user-attachments/assets/7bf58e54-67d1-4e20-989c-5e1ec0828850) which takes audio from my first XLR input port. I guess that's the class compliant interface behaviour. Correct me if I'm wrong, I'm trying to learn about all of this. So, I just wanna be sure that if I go and buy my Roland Rubix, that the device is actually gonna show up the two input lines, and if not that I will know what I will have to do to make it so. Is there gonna be a configuration file required in `alsa-ucm-conf`? Any potentiel change in `snd_usb_audio` kernel module? I'm a bit confused about the details regarding the support of the audio interface devices. Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/459 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ASoC Kernel 6.1: Query regarding dynamic dai link/dapm registration during bootup
by Shalini Manjunatha (Consultant) (QUIC) 08 Nov '24

08 Nov '24

Hi All, On kernel6.1 based Qualcomm target for automotive platform based on Android U, we have a feature early services to launch certain early apps at early init of android bootup cycle. Hence in order to launch early chime sound of audio at early init of android boot up cycle, we need to load all required audio kernel modules at early init stage in order to get sound card registered. We have certain KPI goals for every early apps to get completed within lesser duration so that we can achieve final KPI goal of android boot complete as per the target specification. In this journey, while optimizing the sound card registration through machine driver probe to achieve lower KPI for audio early chime sound, we find that for early chime launch we required only selected pcm front end dai links only. Hence if we keep FE dai link structure in our machine driver with only required items for early chime, overall sound card registrations gets completed within 200msec from the time it starts, which is a very good KPI number to achieve. So we were trying out code change to split overall FE dai link structure into two, one for early chime and another for all other required dai link's, then some how we need to register early chime related dai link during early init so that sound card registration process gets completed quickly, later stage we want to dynamically load all other FE dai links. Could you please help us to suggest on how can we achieve this?. Is there any possible way out to avoid static FE dai link registration for all required ones from machine driver during machine driver probe, instead register only limited list of dai links. Later can we dynamically register the dai links in order to get PCM devices entries created based on use case. Currently we have this API devm_snd_soc_register_card() being called with snd_soc_card parameter which has all possible FE dai links populated at once. Queries: 1) can we register sound card twice during bootup, once with limited dai links and next with all required ones? 2) can we register dai links/dapm dynamically based on use case requriement? 3) If it is possible, could you share any reference from any other vendor who is able to achieve the same?. Regards, Shalini M

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[PATCH RESEND v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 08 Nov '24

08 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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[GIT PULL] ASoC fixes for v6.12-rc6
by Mark Brown 08 Nov '24

08 Nov '24

The following changes since commit 2db63e92186d7201ee1cb2f5af11757c5e5a1020: wcd937x codec fixes (2024-10-29 19:18:48 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc6 for you to fetch changes up to 48b86532c10128cf50c854a90c2d5b1410f4012d: ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits (2024-11-07 15:35:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 A moderately large pile of small changes here, split fairly evenly between fixes and ID additions/quirks and all of it driver specific. ---------------------------------------------------------------- Amelie Delaunay (1): ASoC: stm32: spdifrx: fix dma channel release in stm32_spdifrx_remove Jyri Sarha (1): ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits Kuninori Morimoto (1): MAINTAINERS: Generic Sound Card section Luo Yifan (2): ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate() ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div() Markus Petri (1): ASoC: amd: yc: Support dmic on another model of Lenovo Thinkpad E14 Gen 6 Mingcong Bai (1): ASoC: amd: yc: fix internal mic on Xiaomi Book Pro 14 2022 Shenghao Ding (1): ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip Venkata Prasad Potturu (1): ASoC: SOF: amd: Fix for incorrect DMA ch status register offset MAINTAINERS | 9 +++++++++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/codecs/tas2781-fmwlib.c | 1 + sound/soc/sof/amd/acp.c | 10 +++++++++- sound/soc/sof/sof-client-probes-ipc4.c | 1 + sound/soc/stm/stm32_sai_sub.c | 6 +++--- sound/soc/stm/stm32_spdifrx.c | 2 +- 7 files changed, 38 insertions(+), 5 deletions(-)

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[PATCH RESEND v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 07 Nov '24

07 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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[PATCH v3 1/2] ASoC: dt-bindings: irondevice,sma1307: Add initial DT binding
by Ki-Seok Jo 07 Nov '24

07 Nov '24

This adds the schema binding for the Iron Device SMA1307 Amp Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- .../bindings/sound/irondevice,sma1307.yaml | 54 ++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml new file mode 100644 index 000000000..0bb4ee664 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) %YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1307.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1307 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo(a)irondevice.com> + +description: + SMA1307 boosted digital speaker amplifier + with feedback-loop. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1307a + - irondevice,sma1307aq + description: + If a 'q' is added, it indicated the product is AEC-Q100 + qualified for automotive applications. SMA1307A supports + both WLCSP and QFN packages. However, SMA1307AQ only + supports the QFN package. + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1307a"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; -- 2.39.2

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[PATCH v2] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset
by Venkata Prasad Potturu 07 Nov '24

07 Nov '24

DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: 490be7ba2a01 ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index de3001f5b9bb..95d4762c9d93 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) -- 2.39.2

2 1

[PATCH] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset
by Venkata Prasad Potturu 07 Nov '24

07 Nov '24

DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: bff3c2a4062d ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index ce10d128de25..f7814dadf3ba 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) -- 2.39.2

3 3

topology: Fix strcat() to uninitialized memory in add_define()
by GitHub pull_request - opened 07 Nov '24

07 Nov '24

alsa-project/alsa-utils pull request #277 was opened from rfvirgil: If *defs == NULL the realloc() will be a simple malloc() and return a pointer to uninitialized memory. Doing a strcat() to this is bad, so in that case strcpy() the string into the newly allocated buffer. This fixes the buffer overflow bug, such as: *** buffer overflow detected ***: terminated Program received signal SIGABRT, Aborted. __pthread_kill_implementation (no_tid=0, signo=6, threadid=<optimized out>) at ./nptl/pthread_kill.c:44 warning: 44 ./nptl/pthread_kill.c: No such file or directory (gdb) bt at ../sysdeps/posix/libc_fatal.c:132 at ./debug/fortify_fail.c:24 dest=dest@entry=0x5555555682c0 "hUUU\005PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passth"..., src=src@entry=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., destlen=<optimized out>) at ./debug/strcat_chk.c:34 __src=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., __dest=<optimized out>) at /usr/include/x86_64-linux-gnu/bits/string_fortified.h:130 d=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., defs=<synthetic pointer>) at topology.c:400 Request URL : https://github.com/alsa-project/alsa-utils/pull/277 Patch URL : https://github.com/alsa-project/alsa-utils/pull/277.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR
by schorpp 07 Nov '24

07 Nov '24

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR ALSA versions 1.0.25...current Kernel Driver versions 3.12.40...6.x Hardware: DFI Mainboard MB330-CRM https://www.dfi.com/de/product/index/273#specification Reproducible Always Steps to reproduce and full report: https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… Software crosscheck with CM106 USB Soundbox iec958: works OK. Where to start in the drivers source code to fix it? y tom

1 0

BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is over. The BridgeCast seems to be assigned to the new conf files with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! > _Edits : Spelling/Grammar mistakes_ Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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