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

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 > >

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[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

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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

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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

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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

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[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

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[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

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[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

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[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

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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

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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

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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

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[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

snd_pcm_writei() takes to much time to write data
by GitHub issues - edited 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was edited from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_readi() takes to much time to write data
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was opened from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 17 Dec '24

17 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - removed redundant "&& wov_en" check sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..ecf57a6cb7c3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } - if (is_dmic_enable && wov_en) + if (is_dmic_enable) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

5 5

UCM2: Intel: sof-hda-dsp: Enable Dmic0 pre-processing and SDW speaker post-processing
by GitHub pull_request - edited 16 Dec '24

16 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was edited from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry
by Venkata Prasad Potturu 16 Dec '24

16 Dec '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 0386d765f27a ("ASoC: amd: ps: refactor acp device configuration read logic") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index a7583844f5b4..983d7fa13938 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -374,11 +374,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a { struct acpi_device *pdm_dev; const union acpi_object *obj; + acpi_handle handle; + acpi_integer dmic_status; u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; + bool wov_en, dmic_en; int ret; + /* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/ + wov_en = true; + dmic_en = false; + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: @@ -411,10 +418,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; + dmic_en = true; } + + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; } + if (dmic_en && wov_en) + is_dmic_dev = true; + if (acp_data->is_sdw_config) { ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); if (!ret && acp_data->info.link_mask) -- 2.25.1

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[PATCH v7 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") 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 v7: - add 'select SND_COMPRESS_OFFLOAD' for fixing compile issue with imx_v6_v7_defconfig changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 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 | 2 + 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, 1261 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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