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

alsa-lib-1.2.13 is breaking dssi-1.1.1 because of missing header
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-lib issue #431 was opened from kobboi: What was working with 1.2.12 no longer isn't: ``` /usr/include/alsa/ump_msg.h:1119:65: error: unknown type name ‘size_t’ 1119 | int snd_ump_msg_sysex_expand(const uint32_t *ump, uint8_t *buf, size_t maxlen, | ^~~~~~ /usr/include/alsa/ump_msg.h:13:1: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? 12 | #include <stdint.h> +++ |+#include <stddef.h> 13 | /usr/include/alsa/ump_msg.h:1120:30: error: unknown type name ‘size_t’ 1120 | size_t *filled); | ^~~~~~ /usr/include/alsa/ump_msg.h:1120:30: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? ``` Can this be addressed? Happy holidays! Issue URL : https://github.com/alsa-project/alsa-lib/issues/431 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 30 Dec '24

30 Dec '24

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 910305c11e8a..db50e677c9ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23170,25 +23170,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23198,9 +23215,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

UCM2 macro breaks on updates for Motu 828
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-ucm-conf issue #493 was opened from beatboxa: It appears that every time there is an update, my UCM profile breaks. So that when I boot my machine, I have no sound (unless I manually map things in a pro audio application). This appears to be due to an issue in the SplitPCM macro--specifically, some parsing. Here is the thread describing the creation of the UCM profile I am using, including a patch that I had to apply in order to get things working: #416 And specifically, here is the patch I apply in order to fix it: https://github.com/alsa-project/alsa-ucm-conf/commit/b68aa52acdd2763fedad5e… It involves adding a single hyphen character to the file: ucm2/common/pcm/split.conf Because it is a single character, I have been manually changing the file. It's hard for me to follow that thread--but is this fixed and I am on older versions, or is this still an open item? Or is there a way to automate this so that my audio doesn't break on every update? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/493 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

3 6

[PATCH v4 0/9] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 29 Dec '24

29 Dec '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v4: - Replaced manual regulator toggles with SND_SOC_DAPM_REGULATOR_SUPPLY, both for existing mic bias regulators and for the new headset mic bias regulator - Link to v3: https://lore.kernel.org/r/20240519-midas-wm1811-gpio-jack-v3-0-0c1736144c0e… Changes in v3: - Re-added pipe (|) character to samsung,headset-button-threshold-microvolt description to prevent it from being parsed as a mapping (fixes syntax error) - Split out "ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection" into two patches - Link to v2: https://lore.kernel.org/r/20240508-midas-wm1811-gpio-jack-v2-0-b4d36cd02c6e… Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (9): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Add MCLK2 clock to WM1811 codec config ARM: dts: samsung: exynos4212-tab3: Drop interrupt from WM1811 codec .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 348 +++++++++++++++++---- 4 files changed, 343 insertions(+), 71 deletions(-) --- base-commit: a38297e3fb012ddfa7ce0321a7e5a8daeb1872b6 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

3 12

USB-Audio: ALC4080 - add ASUS ROG Crosshair X870E Hero (USB 0b05:1b7c)
by GitHub pull_request - opened 26 Dec '24

26 Dec '24

alsa-project/alsa-ucm-conf pull request #492 was opened from jetm: https://rog.asus.com/motherboards/rog-crosshair/rog-crosshair-x870e-hero/ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Bug: slab-out-of-bounds in snd_seq_oss_synth_sysex
by Kun Hu 26 Dec '24

26 Dec '24

>> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } If you need any more information, please don't hesitate to let me know. ————— Best regards, Kun Hu > 2024年12月25日 13:37，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： > > Hello, > >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } > > If you fix this issue, please add the following tag to the commit: > Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> > > ————— > Thanks, > Kun Hu > >> 2024年12月24日 19:16，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： >> >> Hello, >> >> When using fuzzer tool to fuzz the latest Linux kernel, the following crash >> was triggered. >> >> HEAD commit: 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8 >> git tree: upstream >> Console output:https://drive.google.com/file/d/17oCyKDW_kNhSW5Bbvm23vnpD1eo0MHFi/vi… >> Kernel config: https://drive.google.com/file/d/1RhT5dFTs6Vx1U71PbpenN7TPtnPoa3NI/view?usp=… >> C reproducer: https://drive.google.com/file/d/177HJht6a7-6F3YLudKb_d4kiPGd1VA_i/view?usp=… >> Syzlang reproducer: https://drive.google.com/file/d/1AuP5UGGc47rEXXPuvjmCKgJ3d0U1P84j/view?usp=… >> >> >> If you fix this issue, please add the following tag to the commit: >> Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> >> >> ================================================================== >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> Write of size 1 at addr ff1100000588e288 by task syz-executor411/824 >> >> CPU: 2 UID: 0 PID: 824 Comm: syz-executor411 Not tainted 6.13.0-rc3 #5 >> Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.13.0-1ubuntu1.1 04/01/2014 >> Call Trace: >> <TASK> >> __dump_stack lib/dump_stack.c:94 [inline] >> dump_stack_lvl+0x116/0x1b0 lib/dump_stack.c:120 >> print_address_description mm/kasan/report.c:378 [inline] >> print_report+0xcf/0x5f0 mm/kasan/report.c:489 >> kasan_report+0x93/0xc0 mm/kasan/report.c:602 >> snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> RIP: 0033:0x7f69b379994d >> Code: c3 e8 97 2a 00 00 0f 1f 80 00 00 00 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 >> RSP: 002b:00007f69b3718d78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 >> RAX: ffffffffffffffda RBX: 00007f69b382d2d0 RCX: 00007f69b379994d >> RDX: 0000000000000008 RSI: 0000000020000140 RDI: 0000000000000003 >> RBP: 00007f69b382d2d8 R08: 0000000000000000 R09: 0000000000000000 >> R10: 0000000000000000 R11: 0000000000000246 R12: 00007f69b382d2dc >> R13: 0000000020000140 R14: 00007f69b37fa008 R15: 000000000000000d >> </TASK> >> >> Allocated by task 823: >> kasan_save_stack+0x24/0x50 mm/kasan/common.c:47 >> kasan_save_track+0x14/0x30 mm/kasan/common.c:68 >> poison_kmalloc_redzone mm/kasan/common.c:377 [inline] >> __kasan_kmalloc+0xaa/0xb0 mm/kasan/common.c:394 >> kmalloc_noprof include/linux/slab.h:901 [inline] >> kzalloc_noprof include/linux/slab.h:1037 [inline] >> snd_seq_oss_synth_sysex+0x1d9/0x6c0 sound/core/seq/oss/seq_oss_synth.c:502 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> >> The buggy address belongs to the object at ff1100000588e200 >> which belongs to the cache kmalloc-192 of size 192 >> The buggy address is located 0 bytes to the right of >> allocated 136-byte region [ff1100000588e200, ff1100000588e288) >> >> The buggy address belongs to the physical page: >> page: refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x588e >> anon flags: 0x100000000000000(node=0|zone=1) >> page_type: f5(slab) >> raw: 0100000000000000 ff1100000103c3c0 ffd4000000162340 dead000000000003 >> raw: 0000000000000000 0000000080100010 00000001f5000000 0000000000000000 >> page dumped because: kasan: bad access detected >> >> Memory state around the buggy address: >> ff1100000588e180: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ff1100000588e200: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 >>> ff1100000588e280: 00 fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >> ^ >> ff1100000588e300: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >> ff1100000588e380: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ================================================================== >> >> --------------- >> thanks, >> Kun Hu > >

1 0

[PATCH 00/21] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 25 Dec '24

25 Dec '24

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (21): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire host wake interrupt handling ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 ++++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 283 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 807 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2840 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

3 23

Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) alsa-info: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - opened 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was opened from River000000: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

[PATCH] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 2

[PATCH V2] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for AMD SoundWire manager power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes Since v1: - Replaced 0x00 with 0 in register write - Updte Commit message drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..ca271a728a65 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

make check fails to compile playmidi1.c: "error: expected identifier or '(' before numeric constant"
by GitHub issues - opened 22 Dec '24

22 Dec '24

alsa-project/alsa-lib issue #430 was opened from 746B61: Getting the latest version from branch master (352cbc5e) and running `./gitcompile` followed by `make check` fails with: ``` Making check in test make[1]: Entering directory '/tmp/alsa-lib-master/test' Making check in . make[2]: Entering directory '/tmp/alsa-lib-master/test' make control pcm pcm_min latency seq seq-ump-example playmidi1 timer rawmidi midiloop umpinfo oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/tmp/alsa-lib-master/test' CC control.o CCLD control CC pcm.o CCLD pcm CC pcm_min.o CCLD pcm_min CC latency.o CCLD latency CC seq.o In file included from seq.c:12: seq-sender.c: In function 'event_sender': seq-sender.c:150:62: warning: variable 'pcm_flag' set but not used [-Wunused-but-set-variable] 150 | int client, port, queue, max, err, v1, v2, time = 0, pcm_flag = 0; | ^~~~~~~~ CCLD seq CC seq-ump-example.o CCLD seq-ump-example CC playmidi1.o In file included from playmidi1.c:48: midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:170:41: note: in expansion of macro 'note_on' 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:171:41: note: in expansion of macro 'note_off' 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:173:41: note: in expansion of macro 'control_change' 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^~~~~~~~~~~~~~ midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:432:41: note: in expansion of macro 'note_on' 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:433:41: note: in expansion of macro 'note_off' 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:438:41: note: in expansion of macro 'control_change' 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^~~~~~~~~~~~~~ In file included from playmidi1.c:49: midifile.c:57:33: error: expected identifier or '(' before numeric constant 57 | #define set_tempo 0x51 | ^~~~ ../include/alsa/ump_msg.h:711:41: note: in expansion of macro 'set_tempo' 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^~~~~~~~~ make[3]: *** [Makefile:622: playmidi1.o] Error 1 make[3]: Leaving directory '/tmp/alsa-lib-master/test' make[2]: *** [Makefile:819: check-am] Error 2 make[2]: Leaving directory '/tmp/alsa-lib-master/test' make[1]: *** [Makefile:669: check-recursive] Error 1 make[1]: Leaving directory '/tmp/alsa-lib-master/test' make: *** [Makefile:414: check-recursive] Error 1 ``` The issue was introduced with commit 07cee0ba. playmidi1.c first [includes](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3… midifile.h from the test directory, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… `#define note_on 0x90`. After commit 07cee0ba, ump_msg.h gets included later, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… ` snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */`. The preprocessor replaces `note_on` with `0x90` and the compilation fails with the error above. Issue URL : https://github.com/alsa-project/alsa-lib/issues/430 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sun4i-codec: add routing for headphones and internal speaker
by GitHub pull_request - opened 21 Dec '24

21 Dec '24

alsa-project/alsa-ucm-conf pull request #491 was opened from tokyovigilante: The sun4i-codec kernel driver covers a large number of Allwinner SoCs. The H616 codec has a single line-out route, which is used in concert with a Toshiba mux chip to send audio to either an internal speaker or headphone jack on a number of Anbernic handheld gaming devices. Add a UCM configuration to allow enabling/disabling the speaker amp depending on whether headphones are in use or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

API memory allocator
by GitHub issues - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-lib issue #429 was opened from sylware: If I recall properly some data returned from the alsa lib API must be free using the C runtime free(). Would it be appropriate to move away from the C allocator to an API install-able memory allocator (like vulkan3D)? Backward compatibility would be achieved since the "default memory allocator" would be the C one, if the C runtime is around. Unless that part of the API is already deprecated (and I should not have used it). Issue URL : https://github.com/alsa-project/alsa-lib/issues/429 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/35] Reorganize kerneldoc parameter names
by Julia Lawall 20 Dec '24

20 Dec '24

Reorganize kerneldoc parameter names to match the parameter order in the function header. The misordered cases were identified using the following Coccinelle semantic patch: // <smpl> @initialize:ocaml@ @@ let parse_doc l = let pieces = List.map String.trim (String.split_on_char '*' l) in let l = String.concat " " pieces in match String.split_on_char ':' l with x::xs -> x | _ -> "" let params ps = List.rev (List.fold_left (fun prev (pm,_) -> let ty = String.trim(Pretty_print_c.string_of_fullType pm.Ast_c.p_type) in if ty = "void" && pm.Ast_c.p_namei = None then prev else let name = match pm.Ast_c.p_namei with Some name -> name | None -> failwith "function parameter has no name" in (String.trim (Pretty_print_c.string_of_name name),ty)::prev) [] ps) @r@ comments c; identifier fn; position p; parameter list ps; type T; @@ T@c fn@p(ps) { ... } @script:ocaml@ p << r.p; c << r.c; (_,ps) << r.ps; @@ let isdoc c ps = List.length ps > 1 && (let c = String.trim c in String.length c > 3 && String.sub c 0 3 = "/**" && String.get c 3 != '*') in let subset l1 l2 = List.for_all (fun x -> List.mem x l2) l1 in let (cb,cm,ca) = List.hd c in match List.rev cb with c::_ when isdoc c ps -> let pieces = String.split_on_char '@' c in (match pieces with _::tl -> let d_names = List.map parse_doc tl in (* check parameters *) let p_names = List.map fst (params ps) in if d_names <> [] && not(d_names = p_names) then begin if List.sort compare d_names = List.sort compare p_names then Coccilib.print_main "out of order" p else if subset d_names p_names then Coccilib.print_main "doc is missing a parameter" p else if subset d_names p_names then Coccilib.print_main "doc has an extra parameter" p end | _ -> ()) | _ -> () // </smpl> --- arch/arm/mach-omap2/prm2xxx_3xxx.c | 1 - arch/powerpc/platforms/ps3/interrupt.c | 2 +- arch/powerpc/platforms/ps3/repository.c | 2 +- drivers/base/firmware_loader/main.c | 2 +- drivers/comedi/drivers/comedi_8254.c | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c | 2 +- drivers/gpu/drm/amd/display/dc/core/dc.c | 2 +- drivers/gpu/drm/drm_gem_framebuffer_helper.c | 3 +-- drivers/gpu/drm/drm_gpuvm.c | 2 +- drivers/gpu/drm/radeon/radeon_ib.c | 2 +- drivers/iommu/iommu.c | 2 +- drivers/leds/leds-gpio-register.c | 2 +- drivers/mfd/atmel-smc.c | 4 ++-- drivers/misc/mei/bus.c | 2 +- drivers/mtd/ubi/eba.c | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt_hwrm.c | 2 +- drivers/net/ethernet/intel/e1000/e1000_hw.c | 2 +- drivers/net/ethernet/intel/i40e/i40e_common.c | 7 +++---- drivers/net/ethernet/intel/ice/ice_common.c | 2 +- drivers/net/ethernet/intel/ixgbe/ixgbe_common.c | 2 +- drivers/nvdimm/dimm_devs.c | 2 +- drivers/pci/hotplug/pci_hotplug_core.c | 2 +- drivers/pinctrl/pinmux.c | 2 +- drivers/slimbus/messaging.c | 2 +- drivers/soc/qcom/qmi_interface.c | 2 +- drivers/soundwire/stream.c | 2 +- drivers/usb/gadget/config.c | 4 ++-- fs/char_dev.c | 2 +- fs/dcache.c | 4 ++-- fs/seq_file.c | 2 +- kernel/audit.c | 2 +- kernel/resource.c | 2 +- kernel/sysctl.c | 1 - kernel/trace/ring_buffer.c | 2 +- lib/lru_cache.c | 2 +- lib/maple_tree.c | 2 +- mm/mmu_notifier.c | 2 +- net/dccp/feat.c | 2 +- net/mac80211/mesh_hwmp.c | 6 +++--- net/mac80211/mesh_pathtbl.c | 10 +++++----- net/socket.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/link.c | 14 +++++++------- net/tipc/msg.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- 45 files changed, 60 insertions(+), 64 deletions(-)

4 5

[alsa-ucm-conf] sof-soundwire: Use correct speaker data port and PCM for cs35l56 bridge
by Maciej Strozek 20 Dec '24

20 Dec '24

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 176f880..2e8c1c6 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -23,8 +23,8 @@ SectionDevice."Speaker" { EnableSequence [ cset "name='cs42l43 Speaker L Input 1' 'ASPRX1'" cset "name='cs42l43 Speaker R Input 1' 'ASPRX2'" - cset "name='cs42l43 ASPTX1 Input' 'DP5RX1'" - cset "name='cs42l43 ASPTX2 Input' 'DP5RX2'" + cset "name='cs42l43 ASPTX1 Input' 'DP6RX1'" + cset "name='cs42l43 ASPTX2 Input' 'DP6RX2'" cset "name='AMPL ASP1 TX1 Source' 'DSP1TX1'" cset "name='AMPL ASP1 TX2 Source' 'None'" @@ -54,7 +54,7 @@ SectionDevice."Speaker" { Value { PlaybackPriority 100 - PlaybackPCM "hw:${CardId},0" + PlaybackPCM "hw:${CardId},2" PlaybackMixer "default:${CardId}" PlaybackMixerElem "Speaker" } -- 2.39.5

2 1

hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[GIT PULL] ASoC fixes for v6.13-rc3
by Mark Brown 20 Dec '24

20 Dec '24

The following changes since commit 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8: Linux 6.13-rc3 (2024-12-15 15:58:23 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc3 for you to fetch changes up to 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A mix of quirks and small fixes, nothing too major anywhere. ---------------------------------------------------------------- Chen-Yu Tsai (2): ASoC: dt-bindings: realtek,rt5645: Fix CPVDD voltage comment ASoC: mediatek: disable buffer pre-allocation Mark Brown (1): ASoC: Intel: sof_sdw: Update DMI matches for Lenovo Peter Ujfalusi (1): ASoC: SOF: Intel: hda-dai: Do not release the link DMA on STOP Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21Q6 and 21Q7 ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21QA and 21QB Shuming Fan (1): ASoC: rt722: add delay time to wait for the calibration procedure Venkata Prasad Potturu (1): ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry .../devicetree/bindings/sound/realtek,rt5645.yaml | 2 +- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++- sound/soc/codecs/rt722-sdca.c | 7 +++++- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++--- .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 ++-- sound/soc/sof/intel/hda-dai.c | 25 ++++++++++++++++------ sound/soc/sof/intel/hda.h | 2 -- 7 files changed, 64 insertions(+), 16 deletions(-)

2 1

[PATCH] ASoC: codecs: ES8326: Reduce pop noise
by Zhang Yi 20 Dec '24

20 Dec '24

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..27736c6933c5 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -771,6 +783,10 @@ static void es8326_jack_button_handler(struct work_struct *work) cur_button = SND_JACK_BTN_0; break; case 0x6f: + if (es8326->version > ES8326_VERSION_B) { + cur_button = SND_JACK_BTN_0; + break; + } case 0x4b: /* button volume up */ cur_button = SND_JACK_BTN_1; @@ -1082,7 +1098,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

1 0

USB-Audio: Add support for TASCAM Model 12
by GitHub pull_request - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-ucm-conf pull request #490 was opened from Unveiled53: Adds support for the TASCAM Model 12 USB multi-track mixer. Couple of things to note: - This profile assumes the "USB AUDIO" setting on the mixer itself is set to "MULTI INPUT". - Under the "MULTI INPUT" mode, this mixer has 12 logical input channels (Last two are a stereo mix) and 10 logical output channels. - Most of the accessible I/O is combo XLR/TRS jacks, so I opted to have the mono capture devices (With the exception of physical inputs 8 and 10 which are TRS only) named as "Mic" with the rest as "Line". Otherwise, the profile is tested and working with the latest USB-Audio profiles. [alsa-info.txt](https://github.com/user-attachments/files/18207924/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

CS42L43 headset mic enable/disable not applied when jack insert state changes
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-ucm-conf issue #489 was opened from rfvirgil: Originally the configuration for sof-soundwire/cs42l43*.conf had the headset mic and internal mics on the same ALSA capture device 4. The Headset Mic enable and disable sequences ran as expected when the headset jack was inserted and removed. But SOF expects that device 4 is the internal mics, and headset mic should be on device 1. So we made this change to move the headset mic to capture device 1: https://github.com/alsa-project/alsa-ucm-conf/commit/30989bd0c2aa3f9f4b6f5e… But now the Headset Mic enable and disable sequences do not run when the headset jack state changes. Seen on multiple systems with Pipewire + alsa-ucm + CS42L43 on sof-soundwire. Example from a system that booted with jack unplugged. The jack was then plugged in. The CS42L43 has reported that the headset mic is present: ``` Simple mixer control 'Headset Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] ``` But the enable sequence has not been applied (DP2TX1 and DP2TX2 should have been set to Decimator 1): ``` Simple mixer control 'cs42l43 DP2TX1 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' Simple mixer control 'cs42l43 DP2TX2 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/489 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0