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Re: regression with SG DMA buf allocations and IOMMU in low-mem
by Kai Vehmanen 14 Nov '22

14 Nov '22

Hi, On Fri, 4 Nov 2022, Takashi Iwai wrote: > On Fri, 04 Nov 2022 15:56:50 +0100, Kai Vehmanen wrote: > > Looking at 5.15 code, the fallback looks very different, but > > in fallback we still use the DMA mapping API via snd_dma_dev_alloc() > > so even if we go to fallback path, mapping is still ok. > > > > Any ideas which way this should be fixed? Given the many changes > > I thought it's better to ask early on the list about this. > > Hrm, that's a tough issue. Basically if dma_alloc_noncontiguous() > fails, it's really out of memory -- at least under the situation where > IOMMU is enabled. The fallback path was introduced for the case where > there is no IOMMU and noncontiguous allocation fails from the > beginning. follow-up to this problem. It turns out there was additional problem with dma_alloc_noncontiguous that led to ENOMEM much more frequently than in the past. This required following: - your rework in 5.16 to sound/core/memalloc.c for x86 DMA SG - IOMMU/DMAR enabled for the device - CONFIG_DMA_REMAP=n (default 'n' on most/all x86 systems before 5.18) ... with above combination, iommu_dma_ops in drivers/iommu/dma-iommu.c were not defined for noncontiguous alloc/free, so the dma_alloc_noncontiguous() in the end went back to plain dma_common_alloc_pages() (single contiguous alloc), which would fail very easily when system was in low-mem condition. In 5.18, this patch was added commit f5ff79fddf0efecca538046b5cc20fb3ded2ec4f Author: Christoph Hellwig <hch(a)lst.de> Date: Sat Feb 26 16:40:21 2022 +0100 dma-mapping: remove CONFIG_DMA_REMAP ... and this basicly fixes the issue again, even when IOMMU is enabled. Br, Kai

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[PATCH] ALSA: memalloc: Allocate more contiguous pages for fallback case
by Takashi Iwai 14 Nov '22

14 Nov '22

Currently the fallback SG allocation tries to allocate each single page, and this tends to result in the reverse order of memory addresses when large space is available at boot, as the kernel takes a free page from the top to the bottom in the zone. The end result looks as if non-contiguous (although it actually is). What's worth is that it leads to an overflow of BDL entries for HD-audio. For avoiding such a problem, this patch modifies the allocation code slightly; now it tries to allocate the larger contiguous chunks as much as possible, then reduces to the smaller chunks only if the allocation failed -- a similar strategy as the existing snd_dma_alloc_pages_fallback() function. Along with the trick, drop the unused address array from snd_dma_sg_fallback object. It was needed in the past when dma_alloc_coherent() was used, but with the standard page allocator, it became superfluous and never referred. Fixes: a8d302a0b770 ("ALSA: memalloc: Revive x86-specific WC page allocations again") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 44 ++++++++++++++++++++++++++----------------- 1 file changed, 27 insertions(+), 17 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index ba095558b6d1..7268304009ad 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -720,7 +720,6 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { struct snd_dma_sg_fallback { size_t count; struct page **pages; - dma_addr_t *addrs; }; static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, @@ -732,38 +731,49 @@ static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) do_free_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); - kvfree(sgbuf->addrs); kfree(sgbuf); } static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) { struct snd_dma_sg_fallback *sgbuf; - struct page **pages; - size_t i, count; + struct page **pagep, *curp; + size_t chunk, npages; + dma_addr_t addr; void *p; bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) return NULL; - count = PAGE_ALIGN(size) >> PAGE_SHIFT; - pages = kvcalloc(count, sizeof(*pages), GFP_KERNEL); - if (!pages) - goto error; - sgbuf->pages = pages; - sgbuf->addrs = kvcalloc(count, sizeof(*sgbuf->addrs), GFP_KERNEL); - if (!sgbuf->addrs) + size = PAGE_ALIGN(size); + sgbuf->count = size >> PAGE_SHIFT; + sgbuf->pages = kvcalloc(sgbuf->count, sizeof(*sgbuf->pages), GFP_KERNEL); + if (!sgbuf->pages) goto error; - for (i = 0; i < count; sgbuf->count++, i++) { - p = do_alloc_pages(dmab->dev.dev, PAGE_SIZE, &sgbuf->addrs[i], wc); - if (!p) - goto error; - sgbuf->pages[i] = virt_to_page(p); + pagep = sgbuf->pages; + chunk = size; + while (size > 0) { + chunk = min(size, chunk); + p = do_alloc_pages(dmab->dev.dev, chunk, &addr, wc); + if (!p) { + if (chunk <= PAGE_SIZE) + goto error; + chunk >>= 1; + chunk = PAGE_SIZE << get_order(chunk); + continue; + } + + size -= chunk; + /* fill pages */ + npages = chunk >> PAGE_SHIFT; + curp = virt_to_page(p); + while (npages--) + *pagep++ = curp++; } - p = vmap(pages, count, VM_MAP, PAGE_KERNEL); + p = vmap(sgbuf->pages, sgbuf->count, VM_MAP, PAGE_KERNEL); if (!p) goto error; dmab->private_data = sgbuf; -- 2.35.3

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[PATCH v2 0/2] ASoC: Intel: avs: DSP recovery and resume fixes
by Cezary Rojewski 14 Nov '22

14 Nov '22

Two fixes that are result of the recent discussions [1][2]. First adds missing locking around snd_pcm_stop() while the second fix sets substream state to DISCONNECTED if any suspend/resume related operation fails so that userspace has means to be aware that something went wrong during said operation. Changes in v2: - __snd_pcm_set_state() replaced direct assignments of PCM state [1]: https://lore.kernel.org/alsa-devel/73e6425f-8e51-e26f-ad83-ccc5df517a43@int… [2]: https://lore.kernel.org/alsa-devel/20221104131244.3920179-1-cezary.rojewski… Cezary Rojewski (2): ASoC: Intel: avs: Lock substream before snd_pcm_stop() ASoC: Intel: avs: Disconnect substream if suspend or resume fails sound/soc/intel/avs/ipc.c | 2 ++ sound/soc/intel/avs/pcm.c | 10 ++++++++-- 2 files changed, 10 insertions(+), 2 deletions(-) -- 2.25.1

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[PATCH v1] sound/pci/hda/patch_realtek: don't call alc_shutup_pins without a spec
by Daniil Tatianin 14 Nov '22

14 Nov '22

alc_shutup_pins always expects the spec to be present, so make sure it is before we call it. Found by Linux Verification Center (linuxtesting.org) with the SVACE static analysis tool. Signed-off-by: Daniil Tatianin <d-tatianin(a)yandex-team.ru> --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 60e3bc124836..2cf4b64971d7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -939,10 +939,12 @@ static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + if (!spec) + return; if (!snd_hda_get_bool_hint(codec, "shutup")) return; /* disabled explicitly by hints */ - if (spec && spec->shutup) + if (spec->shutup) spec->shutup(codec); else alc_shutup_pins(codec); -- 2.25.1

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[PATCH] ALSA: usb-audio: Drop snd_BUG_ON() from snd_usbmidi_output_open()
by Takashi Iwai 12 Nov '22

12 Nov '22

snd_usbmidi_output_open() has a check of the NULL port with snd_BUG_ON(). snd_BUG_ON() was used as this shouldn't have happened, but in reality, the NULL port may be seen when the device gives an invalid endpoint setup at the descriptor, hence the driver skips the allocation. That is, the check itself is valid and snd_BUG_ON() should be dropped from there. Otherwise it's confusing as if it were a real bug, as recently syzbot stumbled on it. Reported-by: syzbot+9abda841d636d86c41da(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/syzbot+9abda841d636d86c41da@syzkaller.appspotmail… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index bbff0923d264..2839f6b6f09b 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1133,10 +1133,8 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) port = &umidi->endpoints[i].out->ports[j]; break; } - if (!port) { - snd_BUG(); + if (!port) return -ENXIO; - } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; -- 2.35.3

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[syzbot] WARNING in snd_usbmidi_output_open
by syzbot 12 Nov '22

12 Nov '22

Hello, syzbot found the following issue on: HEAD commit: f8f60f322f06 Add linux-next specific files for 20221111 git tree: linux-next console+strace: https://syzkaller.appspot.com/x/log.txt?x=12fe828e880000 kernel config: https://syzkaller.appspot.com/x/.config?x=85ba52c07cd97289 dashboard link: https://syzkaller.appspot.com/bug?extid=9abda841d636d86c41da compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=160e91fa880000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=161d3e69880000 Downloadable assets: disk image: https://storage.googleapis.com/syzbot-assets/6008df424195/disk-f8f60f32.raw… vmlinux: https://storage.googleapis.com/syzbot-assets/394340525f66/vmlinux-f8f60f32.… kernel image: https://storage.googleapis.com/syzbot-assets/b13604a3343a/bzImage-f8f60f32.… IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+9abda841d636d86c41da(a)syzkaller.appspotmail.com ------------[ cut here ]------------ BUG? WARNING: CPU: 0 PID: 5267 at sound/usb/midi.c:1137 snd_usbmidi_output_open+0xd8/0x410 sound/usb/midi.c:1137 Modules linked in: CPU: 0 PID: 5267 Comm: syz-executor895 Not tainted 6.1.0-rc4-next-20221111-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 10/26/2022 RIP: 0010:snd_usbmidi_output_open+0xd8/0x410 sound/usb/midi.c:1137 Code: 3c 02 00 0f 85 03 03 00 00 4d 8b be 10 02 00 00 4d 85 ff 0f 85 c8 00 00 00 e8 d4 92 25 fa 48 c7 c7 a0 6e f2 8a e8 02 da 04 02 <0f> 0b 41 bc fa ff ff ff e8 bb 92 25 fa 44 89 e0 48 83 c4 08 5b 5d RSP: 0018:ffffc90003b7f3a0 EFLAGS: 00010282 RAX: 0000000000000000 RBX: ffff888146c8d000 RCX: 0000000000000000 RDX: ffff8880245d3a80 RSI: ffffffff81624e68 RDI: fffff5200076fe66 RBP: ffff88801ed9c900 R08: 0000000000000005 R09: 0000000000000000 R10: 0000000080000000 R11: 0000000000000000 R12: ffff88814017cc00 R13: 0000000000000000 R14: ffff88802b1d5000 R15: 0000000000000000 FS: 00005555574d4300(0000) GS:ffff8880b9a00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f6ed57c3670 CR3: 0000000076149000 CR4: 00000000003506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <TASK> open_substream+0x41d/0x8c0 sound/core/rawmidi.c:344 rawmidi_open_priv+0x591/0x6f0 sound/core/rawmidi.c:394 snd_rawmidi_kernel_open+0x1b5/0x270 sound/core/rawmidi.c:429 midisynth_use+0xee/0x270 sound/core/seq/seq_midi.c:215 subscribe_port sound/core/seq/seq_ports.c:412 [inline] check_and_subscribe_port+0x89a/0xb80 sound/core/seq/seq_ports.c:495 snd_seq_port_connect+0x382/0x540 sound/core/seq/seq_ports.c:581 snd_seq_ioctl_subscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1490 snd_seq_kernel_client_ctl+0x102/0x1c0 sound/core/seq/seq_clientmgr.c:2367 snd_seq_oss_midi_open+0x582/0x6e0 sound/core/seq/oss/seq_oss_midi.c:361 snd_seq_oss_synth_setup_midi+0x12d/0x530 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x8b8/0xa70 sound/core/seq/oss/seq_oss_init.c:260 odev_open+0x6c/0x90 sound/core/seq/oss/seq_oss.c:128 soundcore_open+0x44e/0x620 sound/sound_core.c:593 chrdev_open+0x266/0x770 fs/char_dev.c:414 do_dentry_open+0x6cc/0x13f0 fs/open.c:882 do_open fs/namei.c:3557 [inline] path_openat+0x1bbc/0x2a50 fs/namei.c:3713 do_filp_open+0x1b6/0x400 fs/namei.c:3740 do_sys_openat2+0x16d/0x4c0 fs/open.c:1310 do_sys_open fs/open.c:1326 [inline] __do_sys_openat fs/open.c:1342 [inline] __se_sys_openat fs/open.c:1337 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1337 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x63/0xcd RIP: 0033:0x7f496367ae39 Code: 28 00 00 00 75 05 48 83 c4 28 c3 e8 b1 14 00 00 90 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 c0 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007ffc99e01aa8 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 00000000000f4240 RCX: 00007f496367ae39 RDX: 0000000000020983 RSI: 0000000020000080 RDI: ffffffffffffff9c RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000001 R10: 0000000000000000 R11: 0000000000000246 R12: 000000000000d066 R13: 00007ffc99e01abc R14: 00007ffc99e01ad0 R15: 00007ffc99e01ac0 </TASK> --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

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[GIT PULL] ASoC fixes for v6.2-rc4
by Mark Brown 12 Nov '22

12 Nov '22

The following changes since commit e9441675edc1bb8dbfadacf68aafacca60d65a25: ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() (2022-10-21 13:04:14 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.2-rc4 for you to fetch changes up to 7d945b046be3d2605dbb1806e73095aadd7ae129: ASoC: stm32: dfsdm: manage cb buffers cleanup (2022-11-09 17:56:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.1 A relatively large collection of fixes and new platform quirks here, they're all fairly minor though - the widest possible impact is the fix to the use of prefixes on regulator names which would have broken any device that integrates regulators with DAPM and was used in a system where it had a name prefix assigning to it. ---------------------------------------------------------------- Brent Mendelsohn (1): ASoC: amd: yc: Add Alienware m17 R5 AMD into DMI table Chen Zhongjin (2): ASoC: core: Fix use-after-free in snd_soc_exit() ASoC: soc-utils: Remove __exit for snd_soc_util_exit() Hans de Goede (1): ASoC: Intel: bytcht_es8316: Add quirk for the Nanote UMPC-01 Jason Montleon (2): ASoC: rt5514: fix legacy dai naming ASoC: rt5677: fix legacy dai naming Kai Vehmanen (1): ASoC: SOF: ipc3-topology: use old pipeline teardown flow with SOF2.1 and older Maarten Zanders (1): ASoC: fsl_asrc fsl_esai fsl_sai: allow CONFIG_PM=N Martin Povišer (3): ASoC: tas2770: Fix set_tdm_slot in case of single slot ASoC: tas2764: Fix set_tdm_slot in case of single slot ASoC: tas2780: Fix set_tdm_slot in case of single slot Olivier Moysan (1): ASoC: stm32: dfsdm: manage cb buffers cleanup Paul Cercueil (1): ASoC: dapm: Don't use prefix for regulator name Peter Ujfalusi (1): ASoC: SOF: topology: No need to assign core ID if token parsing failed Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: add ES83x6 support to IceLake ASoC: hda: intel-dsp-config: add ES83x6 quirk for IceLake Tzung-Bi Shih (1): MAINTAINERS: update Tzung-Bi's email address Zhu Ning (1): ASoC: sof_es8336: reduce pop noise on speaker .mailmap | 1 + .../bindings/sound/google,cros-ec-codec.yaml | 2 +- .../devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- include/sound/sof/info.h | 4 ++ sound/hda/intel-dsp-config.c | 5 ++ sound/soc/amd/yc/acp6x-mach.c | 7 +++ sound/soc/codecs/rt5514-spi.c | 15 +++--- sound/soc/codecs/rt5677-spi.c | 19 +++---- sound/soc/codecs/tas2764.c | 19 +++---- sound/soc/codecs/tas2770.c | 20 +++----- sound/soc/codecs/tas2780.c | 19 +++---- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 7 +++ sound/soc/intel/boards/sof_es8336.c | 60 +++++++++++++++++----- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 13 +++++ sound/soc/soc-core.c | 17 +++++- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-utils.c | 2 +- sound/soc/sof/ipc3-topology.c | 15 ++++-- sound/soc/sof/topology.c | 20 ++++---- sound/soc/stm/stm32_adfsdm.c | 11 ++++ 24 files changed, 173 insertions(+), 95 deletions(-)

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[PATCH] ALSA: memalloc: Try dma_alloc_noncontiguous() at first
by Takashi Iwai 12 Nov '22

12 Nov '22

The latest fix for the non-contiguous memalloc helper changed the allocation method for a non-IOMMU system to use only the fallback allocator. This should have worked, but it caused a problem sometimes when too many non-contiguous pages are allocated that can't be treated by HD-audio controller. As a quirk workaround, go back to the original strategy: use dma_alloc_noncontiguous() at first, and apply the fallback only when it fails, but only for non-IOMMU case. We'll need a better fix in the fallback code as well, but this workaround should paper over most cases. Fixes: 9736a325137b ("ALSA: memalloc: Don't fall back for SG-buffer with IOMMU") Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Link: https://lore.kernel.org/r/CAHk-=wgSH5ubdvt76gNwa004ooZAEJL_1Q-Fyw5M2FDdqL==… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6a81aaab25ab..ba095558b6d1 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -542,8 +542,10 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) struct sg_table *sgt; void *p; + sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, + DEFAULT_GFP, 0); #ifdef CONFIG_SND_DMA_SGBUF - if (!get_dma_ops(dmab->dev.dev)) { + if (!sgt && !get_dma_ops(dmab->dev.dev)) { if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; else @@ -551,9 +553,6 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) return snd_dma_sg_fallback_alloc(dmab, size); } #endif - - sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, - DEFAULT_GFP, 0); if (!sgt) return NULL; -- 2.35.3

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Build failure with upcoming Clang 16 (-Wincompatible-function-pointer-types)
by GitHub issues - opened 12 Nov '22

12 Nov '22

alsa-project/alsa-tools issue #12 was opened from thesamesam: Clang 16 will make -Wincompatible-function-pointer-types an error by default. You can emulate this error using: * GCC via -Werror=incompatible-pointer-types (GCC lacks a more specific warning for just func. ptrs) * Clang <16 via -Werror=incompatible-function-pointer-types Originally reported downstream in Gentoo at https://bugs.gentoo.org/880997. ``` echomixer.c:2108:7: error: incompatible function pointer types passing 'void (GtkWidget *, gpointer)' (aka 'void (struct _GtkWidget *, void *)') to parameter of type 'GCallback' (aka 'void (*)(void)') [-Werror,-Wincompatible-function-pointer-types] ...gtk_signal_connect(GTK_OBJECT(menuitem), "activate", Digital_mode_activate, (gpointer)(long)i); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ /usr/include/gtk-2.0/gtk/gtksignal.h:51:47: note: expanded from macro 'gtk_signal_connect' gtk_signal_connect_full ((object), (name), (func), NULL, (func_data)... ^~~~~~ /usr/include/gtk-2.0/gtk/gtksignal.h:123:23: note: passing argument to parameter 'func' here GCallback func, ^ [...] ``` Full build log: https://bugs.gentoo.org/attachment.cgi?id=831477 Issue URL : https://github.com/alsa-project/alsa-tools/issues/12 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH v2] ASoC: codecs: wsa883x: use correct header file
by Randy Dunlap 11 Nov '22

11 Nov '22

Fix build errors when GPIOLIB is not set/enabled: ../sound/soc/codecs/wsa883x.c: In function 'wsa883x_probe': ../sound/soc/codecs/wsa883x.c:1394:25: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] wsa883x->sd_n = devm_gpiod_get_optional(&pdev->dev, "powerdown", ../sound/soc/codecs/wsa883x.c:1395:49: error: 'GPIOD_FLAGS_BIT_NONEXCLUSIVE' undeclared (first use in this function) GPIOD_FLAGS_BIT_NONEXCLUSIVE); ../sound/soc/codecs/wsa883x.c:1414:9: error: implicit declaration of function 'gpiod_direction_output'; did you mean 'gpio_direction_output'? [-Werror=implicit-function-declaration] gpiod_direction_output(wsa883x->sd_n, 1); Fixes: 43b8c7dc85a1 ("ASoC: codecs: add wsa883x amplifier support") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Banajit Goswami <bgoswami(a)quicinc.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> --- v2: add the correct header file instead of messing with GPIOLIB in Kconfig sound/soc/codecs/wsa883x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -- a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -7,7 +7,7 @@ #include <linux/debugfs.h> #include <linux/delay.h> #include <linux/device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h>

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[PATCH v2 1/2] ASoC: dt-bindings: qcom, wsa883x: Use correct SD_N polarity
by Krzysztof Kozlowski 11 Nov '22

11 Nov '22

Use correct polarity in example and powerdown-gpios description. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes since v1: 1. New patch. --- Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml index 6113f65f2990..99f9c10bbc83 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -23,7 +23,7 @@ properties: maxItems: 1 powerdown-gpios: - description: GPIO spec for Powerdown/Shutdown line to use + description: GPIO spec for Powerdown/Shutdown line to use (pin SD_N) maxItems: 1 vdd-supply: @@ -47,6 +47,8 @@ additionalProperties: false examples: - | + #include <dt-bindings/gpio/gpio.h> + soundwire-controller@3250000 { #address-cells = <2>; #size-cells = <0>; @@ -55,7 +57,7 @@ examples: speaker@0,1 { compatible = "sdw10217020200"; reg = <0 1>; - powerdown-gpios = <&tlmm 1 0>; + powerdown-gpios = <&tlmm 1 GPIO_ACTIVE_LOW>; vdd-supply = <&vreg_s10b_1p8>; #thermal-sensor-cells = <0>; #sound-dai-cells = <0>; @@ -64,7 +66,7 @@ examples: speaker@0,2 { compatible = "sdw10217020200"; reg = <0 2>; - powerdown-gpios = <&tlmm 89 0>; + powerdown-gpios = <&tlmm 89 GPIO_ACTIVE_LOW>; vdd-supply = <&vreg_s10b_1p8>; #thermal-sensor-cells = <0>; #sound-dai-cells = <0>; -- 2.34.1

3 4

[PATCH 0/4] ASoC: Intel: add new Dell devices support
by Bard Liao 11 Nov '22

11 Nov '22

Add 2 Dell SKU support. Gongjun Song (4): ASoC: Intel: sof_sdw: Add support for SKU 0C10 product ASoC: Intel: soc-acpi: add SKU 0C10 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0C40 product ASoC: Intel: soc-acpi: add SKU 0C40 SoundWire configuration sound/soc/intel/boards/sof_sdw.c | 21 ++++ .../intel/common/soc-acpi-intel-rpl-match.c | 109 ++++++++++++++++++ 2 files changed, 130 insertions(+) -- 2.25.1

2 5

[PATCH] ASoC: audio-graph-card2: remove Experimental announce
by Kuninori Morimoto 11 Nov '22

11 Nov '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Audio Graph Card2 has been announcing "Audio Graph Card2 is still under Experimental stage", but it is time to remove it. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card2.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 8ac6df645ee6..0b0fec9ada51 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1271,9 +1271,6 @@ int audio_graph2_parse_of(struct asoc_simple_priv *priv, struct device *dev, if (ret < 0) dev_err_probe(dev, ret, "parse error\n"); - if (ret == 0) - dev_warn(dev, "Audio Graph Card2 is still under Experimental stage\n"); - return ret; } EXPORT_SYMBOL_GPL(audio_graph2_parse_of); -- 2.25.1

2 1

[PATCH] ASoC: qcom: q6prm: Correct module description
by Krzysztof Kozlowski 11 Nov '22

11 Nov '22

The description was copied from APM driver. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/q6prm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index cda33ded29be..8aa1a213bfb7 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -247,5 +247,5 @@ static gpr_driver_t prm_driver = { }; module_gpr_driver(prm_driver); -MODULE_DESCRIPTION("Audio Process Manager"); +MODULE_DESCRIPTION("Q6 Proxy Resource Manager"); MODULE_LICENSE("GPL"); -- 2.34.1

2 1

[PATCH 0/5] ASoC: Intel: add more ACPI tables/quirks for RPL/MTL
by Pierre-Louis Bossart 11 Nov '22

11 Nov '22

Add support for two RaptorLake SoundWire SKUs and one test board for MeteorLake. There will be additional RaptorLake SKUs shared when validation is complete. Gongjun Song (4): ASoC: Intel: sof_sdw: Add support for SKU 0C10 product ASoC: Intel: soc-acpi: add SKU 0C10 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0C40 product ASoC: Intel: soc-acpi: add SKU 0C40 SoundWire configuration Yong Zhi (1): ASoC: Intel: soc-acpi: add MTL AIC SoundWire configurations sound/soc/intel/boards/sof_sdw.c | 21 ++++ .../intel/common/soc-acpi-intel-mtl-match.c | 59 ++++++++++ .../intel/common/soc-acpi-intel-rpl-match.c | 109 ++++++++++++++++++ 3 files changed, 189 insertions(+) -- 2.34.1

2 6

[PATCH] ASoC: soc-pcm: Don't zero TDM masks in __soc_pcm_open()
by Richard Fitzgerald 11 Nov '22

11 Nov '22

The DAI tx_mask and rx_mask are set by snd_soc_dai_set_tdm_slot() and used by later code that depends on the TDM settings. So __soc_pcm_open() should not be obliterating those mask values. The code in __soc_pcm_hw_params() uses these masks to calculate the active channels so that only the AIF_IN/AIF_OUT widgets for the active TDM slots are enabled. The zeroing of the masks in __soc_pcm_open() disables this functionality so all AIF widgets were enabled even for channels that are not assigned to a TDM slot. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 2e5894d73789 ("ASoC: pcm: Add support for DAI multicodec") --- sound/soc/soc-pcm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d8e4677f3002..493f003273d0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -822,11 +822,6 @@ static int __soc_pcm_open(struct snd_soc_pcm_runtime *rtd, ret = snd_soc_dai_startup(dai, substream); if (ret < 0) goto err; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->tx_mask = 0; - else - dai->rx_mask = 0; } /* Dynamic PCM DAI links compat checks use dynamic capabilities */ -- 2.30.2

2 1

[PATCH 4/4] ASoC: Intel: sof_sdw_rt1316: add BQ params for the Dell models
by shumingf＠realtek.com 11 Nov '22

11 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> The Dell SKU 0B00/0B01/0AFE/0AFF model needs the BQ params for the tweeter/woofer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reviewed-by: Pierre-Louis Bossart pierre-louis.bossart(a)linux.intel.com --- sound/soc/intel/boards/sof_sdw.c | 1 + .../intel/boards/sof_sdw_amp_coeff_tables.h | 138 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 1 + sound/soc/intel/boards/sof_sdw_rt1316.c | 119 +++++++++++++++ 4 files changed, 259 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a3bbdc261a4a..949adfdd0e3c 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -518,6 +518,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .direction = {true, true}, .dai_name = "rt1316-aif", .init = sof_sdw_rt1316_init, + .exit = sof_sdw_rt1316_exit, .codec_type = SOF_SDW_CODEC_TYPE_AMP, }, { diff --git a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h index bb90a5922fd2..82230fbb5b05 100644 --- a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h +++ b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h @@ -9,6 +9,7 @@ #define SND_SOC_SOF_SDW_AMP_COEFF_H #define RT1308_MAX_BQ_REG 480 +#define RT1316_MAX_BQ_REG 580 static const u8 dell_0a5d_bq_params[] = { 0xb0, 0xc5, 0x00, /* address: 0xc5b0; data: 0x00 */ @@ -159,4 +160,141 @@ static const u8 dell_0a5d_bq_params[] = { 0x50, 0xc5, 0x01, }; +static const u8 dell_0b00_bq_params[] = { + 0x03, 0xc2, 0x00, + 0x04, 0xc2, 0xb2, + 0x05, 0xc2, 0xe0, + 0x06, 0xc2, 0x3a, + 0x07, 0xc2, 0x01, + 0x08, 0xc2, 0x65, + 0x09, 0xc2, 0xc0, + 0x0a, 0xc2, 0x75, + 0x0b, 0xc2, 0x00, + 0x0c, 0xc2, 0xb2, + 0x0d, 0xc2, 0xe0, + 0x0e, 0xc2, 0x3a, + 0x0f, 0xc2, 0xf7, + 0x10, 0xc2, 0x4d, + 0x11, 0xc2, 0x5b, + 0x12, 0xc2, 0xe9, + 0x13, 0xc2, 0x03, + 0x14, 0xc2, 0x7e, + 0x15, 0xc2, 0x25, + 0x16, 0xc2, 0x01, + 0x17, 0xc2, 0x07, + 0x18, 0xc2, 0xfd, + 0x19, 0xc2, 0x15, + 0x1a, 0xc2, 0x04, + 0x1b, 0xc2, 0xf0, + 0x1c, 0xc2, 0x05, + 0x1d, 0xc2, 0xd5, + 0x1e, 0xc2, 0xf7, + 0x1f, 0xc2, 0x07, + 0x20, 0xc2, 0xfd, + 0x21, 0xc2, 0x15, + 0x22, 0xc2, 0x04, + 0x23, 0xc2, 0xf0, + 0x24, 0xc2, 0x05, + 0x25, 0xc2, 0xd8, + 0x26, 0xc2, 0x17, + 0x27, 0xc2, 0x07, + 0x28, 0xc2, 0xfa, + 0x29, 0xc2, 0x2c, + 0x2a, 0xc2, 0x29, + 0x2b, 0xc2, 0x07, + 0x2c, 0xc2, 0x74, + 0x2d, 0xc2, 0xe0, + 0x2e, 0xc2, 0x33, + 0x2f, 0xc2, 0xf1, + 0x30, 0xc2, 0x16, + 0x31, 0xc2, 0x3f, + 0x32, 0xc2, 0x9b, + 0x33, 0xc2, 0x07, + 0x34, 0xc2, 0x74, + 0x35, 0xc2, 0xe0, + 0x36, 0xc2, 0x33, + 0x37, 0xc2, 0xf1, + 0x38, 0xc2, 0x29, + 0x39, 0xc2, 0xb0, + 0x3a, 0xc2, 0x4d, + 0x3b, 0xc2, 0x06, + 0x3c, 0xc2, 0xfd, + 0x3d, 0xc2, 0x31, + 0x3e, 0xc2, 0x18, + 0x3f, 0xc2, 0x07, + 0x40, 0xc2, 0xfd, + 0x41, 0xc2, 0x15, + 0x42, 0xc2, 0x04, + 0x43, 0xc2, 0xf0, + 0x44, 0xc2, 0x05, + 0x45, 0xc2, 0xd5, + 0x46, 0xc2, 0xf7, + 0x47, 0xc2, 0x07, + 0x48, 0xc2, 0xfd, + 0x49, 0xc2, 0x15, + 0x4a, 0xc2, 0x04, + 0x4b, 0xc2, 0xf0, + 0x4c, 0xc2, 0x05, + 0x4d, 0xc2, 0xd8, + 0x4e, 0xc2, 0x17, + 0x4f, 0xc2, 0x07, + 0x50, 0xc2, 0xfa, + 0x51, 0xc2, 0x2c, + 0x52, 0xc2, 0x29, + 0x0b, 0xc0, 0x30, + 0x80, 0xc3, 0x13, + 0x81, 0xc3, 0x88, + 0x82, 0xc3, 0x17, + 0x83, 0xc3, 0x70, + 0x84, 0xc3, 0x00, + 0x85, 0xc3, 0x00, + 0x86, 0xc3, 0xff, + 0x87, 0xc3, 0xee, + 0x88, 0xc3, 0x02, + 0x92, 0xc3, 0x00, + 0x93, 0xc3, 0x14, + 0x94, 0xc3, 0x00, + 0x95, 0xc3, 0x14, + 0x96, 0xc3, 0x00, + 0x97, 0xc3, 0x00, + 0x98, 0xc3, 0x00, + 0x99, 0xc3, 0x00, + 0x9a, 0xc3, 0x01, + 0x89, 0xc3, 0x03, + 0x8a, 0xc3, 0xe8, + 0x8b, 0xc3, 0x03, + 0x8c, 0xc3, 0xb6, + 0x8d, 0xc3, 0x00, + 0x8e, 0xc3, 0x00, + 0x8f, 0xc3, 0xff, + 0x90, 0xc3, 0xee, + 0x91, 0xc3, 0x01, + 0x9b, 0xc3, 0x00, + 0x9c, 0xc3, 0x14, + 0x9d, 0xc3, 0x00, + 0x9e, 0xc3, 0x14, + 0x9f, 0xc3, 0x00, + 0xa0, 0xc3, 0x00, + 0xa1, 0xc3, 0x00, + 0xa2, 0xc3, 0x00, + 0xa3, 0xc3, 0x01, + 0x61, 0xc2, 0x08, + 0x62, 0xc2, 0x00, + 0x63, 0xc2, 0x00, + 0x64, 0xc2, 0x00, + 0x65, 0xc2, 0x08, + 0x66, 0xc2, 0x00, + 0x67, 0xc2, 0x00, + 0x68, 0xc2, 0x00, + 0x69, 0xc2, 0x08, + 0x6a, 0xc2, 0x00, + 0x6b, 0xc2, 0x00, + 0x6c, 0xc2, 0x00, + 0x6d, 0xc2, 0x08, + 0x6e, 0xc2, 0x00, + 0x6f, 0xc2, 0x00, + 0x70, 0xc2, 0x00, + 0x00, 0xc2, 0xc0, +}; + #endif diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 772af23e7497..bac93fdc8d82 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -141,6 +141,7 @@ int sof_sdw_rt1316_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); +int sof_sdw_rt1316_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT715 support */ int sof_sdw_rt715_init(struct snd_soc_card *card, diff --git a/sound/soc/intel/boards/sof_sdw_rt1316.c b/sound/soc/intel/boards/sof_sdw_rt1316.c index 58194b380232..f6bbea0d3810 100644 --- a/sound/soc/intel/boards/sof_sdw_rt1316.c +++ b/sound/soc/intel/boards/sof_sdw_rt1316.c @@ -11,7 +11,83 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include <sound/soc-dapm.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/dmi.h> #include "sof_sdw_common.h" +#include "sof_sdw_amp_coeff_tables.h" + +struct rt1316_platform_data { + const unsigned char *bq_params; + const unsigned int bq_params_cnt; +}; + +static const struct rt1316_platform_data dell_0b00_platform_data = { + .bq_params = dell_0b00_bq_params, + .bq_params_cnt = ARRAY_SIZE(dell_0b00_bq_params), +}; + +static const struct dmi_system_id dmi_platform_data[] = { + /* AlderLake devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B00") + }, + .driver_data = (void *)&dell_0b00_platform_data, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B01") + }, + .driver_data = (void *)&dell_0b00_platform_data, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFF") + }, + .driver_data = (void *)&dell_0b00_platform_data, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFE") + }, + .driver_data = (void *)&dell_0b00_platform_data, + }, +}; + +static int rt1316_add_device_props(struct device *sdw_dev) +{ + struct property_entry props[3] = {}; + struct fwnode_handle *fwnode; + const struct dmi_system_id *dmi_data; + const struct rt1316_platform_data *pdata; + unsigned char params[RT1316_MAX_BQ_REG]; + int ret; + + dmi_data = dmi_first_match(dmi_platform_data); + if (!dmi_data) + return 0; + + pdata = dmi_data->driver_data; + memcpy(&params, pdata->bq_params, sizeof(unsigned char) * pdata->bq_params_cnt); + + props[0] = PROPERTY_ENTRY_U8_ARRAY("realtek,bq-params", params); + props[1] = PROPERTY_ENTRY_U32("realtek,bq-params-cnt", pdata->bq_params_cnt); + + fwnode = fwnode_create_software_node(props, NULL); + if (IS_ERR(fwnode)) + return PTR_ERR(fwnode); + + ret = device_add_software_node(sdw_dev, to_software_node(fwnode)); + + fwnode_handle_put(fwnode); + + return ret; +} static const struct snd_soc_dapm_widget rt1316_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), @@ -89,12 +165,33 @@ static int all_spk_init(struct snd_soc_pcm_runtime *rtd) return second_spk_init(rtd); } +int sof_sdw_rt1316_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->amp_dev1) { + device_remove_software_node(ctx->amp_dev1); + put_device(ctx->amp_dev1); + } + + if (ctx->amp_dev2) { + device_remove_software_node(ctx->amp_dev2); + put_device(ctx->amp_dev2); + } + + return 0; +} + int sof_sdw_rt1316_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) { + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct device *sdw_dev1, *sdw_dev2; + int ret; + /* Count amp number and do init on playback link only. */ if (!playback) return 0; @@ -104,6 +201,28 @@ int sof_sdw_rt1316_init(struct snd_soc_card *card, dai_links->init = first_spk_init; if (info->amp_num == 2) { + sdw_dev1 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name); + if (!sdw_dev1) + return -EPROBE_DEFER; + + ret = rt1316_add_device_props(sdw_dev1); + if (ret < 0) { + put_device(sdw_dev1); + return ret; + } + ctx->amp_dev1 = sdw_dev1; + + sdw_dev2 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[1].name); + if (!sdw_dev2) + return -EPROBE_DEFER; + + ret = rt1316_add_device_props(sdw_dev2); + if (ret < 0) { + put_device(sdw_dev2); + return ret; + } + ctx->amp_dev2 = sdw_dev2; + /* * if two 1316s are in one dai link, the init function * in this dai link will be first set for the first speaker, -- 2.38.1

2 1

[PATCH 3/4] ASoC: Intel: sof_sdw_rt1308: add BQ params for the Dell models
by shumingf＠realtek.com 11 Nov '22

11 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> The Dell SKU 0A5D/0A5E/0990/098F model needs the BQ params for the tweeter/woofer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reviewed-by: Pierre-Louis Bossart pierre-louis.bossart(a)linux.intel.com --- sound/soc/intel/boards/sof_sdw.c | 1 + .../intel/boards/sof_sdw_amp_coeff_tables.h | 162 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 2 + sound/soc/intel/boards/sof_sdw_rt1308.c | 120 +++++++++++++ 4 files changed, 285 insertions(+) create mode 100644 sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index ee9857dc3135..a3bbdc261a4a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -510,6 +510,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt1308-aif", .ops = &sof_sdw_rt1308_i2s_ops, .init = sof_sdw_rt1308_init, + .exit = sof_sdw_rt1308_exit, .codec_type = SOF_SDW_CODEC_TYPE_AMP, }, { diff --git a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h new file mode 100644 index 000000000000..bb90a5922fd2 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h @@ -0,0 +1,162 @@ +/* SPDX-License-Identifier: GPL-2.0-only + */ + +/* + * sof_sdw_amp_coeff_tables.h - related coefficients for amplifier parameters + */ + +#ifndef SND_SOC_SOF_SDW_AMP_COEFF_H +#define SND_SOC_SOF_SDW_AMP_COEFF_H + +#define RT1308_MAX_BQ_REG 480 + +static const u8 dell_0a5d_bq_params[] = { + 0xb0, 0xc5, 0x00, /* address: 0xc5b0; data: 0x00 */ + 0xb1, 0xc5, 0x32, + 0xb2, 0xc5, 0x44, + 0xb3, 0xc5, 0x19, + 0xc0, 0xc5, 0x04, + 0xc1, 0xc5, 0x00, + 0xc2, 0xc5, 0x00, + 0xc3, 0xc5, 0x00, + 0xd0, 0xc5, 0x02, + 0xd1, 0xc5, 0x00, + 0xd2, 0xc5, 0x00, + 0xd3, 0xc5, 0x00, + 0xe0, 0xc5, 0x01, + 0xe1, 0xc5, 0xe8, + 0xe2, 0xc5, 0x5f, + 0xe3, 0xc5, 0x8a, + 0xf0, 0xc5, 0x1f, + 0xf1, 0xc5, 0x4e, + 0xf2, 0xc5, 0x90, + 0xf3, 0xc5, 0x11, + 0x50, 0xc6, 0x01, + 0x51, 0xc6, 0xff, + 0x52, 0xc6, 0x45, + 0x53, 0xc6, 0x41, + 0x60, 0xc6, 0x1c, + 0x61, 0xc6, 0x00, + 0x62, 0xc6, 0x00, + 0x63, 0xc6, 0x00, + 0x70, 0xc6, 0x02, + 0x71, 0xc6, 0x00, + 0x72, 0xc6, 0x00, + 0x73, 0xc6, 0x00, + 0x80, 0xc6, 0x03, + 0x81, 0xc6, 0xfe, + 0x82, 0xc6, 0x89, + 0x83, 0xc6, 0xfa, + 0x90, 0xc6, 0x1e, + 0x91, 0xc6, 0x01, + 0x92, 0xc6, 0x74, + 0x93, 0xc6, 0xf6, + 0x00, 0xc6, 0x01, + 0x01, 0xc6, 0xd9, + 0x02, 0xc6, 0xfb, + 0x03, 0xc6, 0xc4, + 0x10, 0xc6, 0x1c, + 0x11, 0xc6, 0x00, + 0x12, 0xc6, 0x00, + 0x13, 0xc6, 0x00, + 0x20, 0xc6, 0x02, + 0x21, 0xc6, 0x00, + 0x22, 0xc6, 0x00, + 0x23, 0xc6, 0x00, + 0x30, 0xc6, 0x03, + 0x31, 0xc6, 0xaf, + 0x32, 0xc6, 0x23, + 0x33, 0xc6, 0xcb, + 0x40, 0xc6, 0x1e, + 0x41, 0xc6, 0x47, + 0x42, 0xc6, 0x34, + 0x43, 0xc6, 0xba, + 0xa0, 0xc6, 0x01, + 0xa1, 0xc6, 0xff, + 0xa2, 0xc6, 0x45, + 0xa3, 0xc6, 0x41, + 0xb0, 0xc6, 0x1c, + 0xb1, 0xc6, 0x00, + 0xb2, 0xc6, 0x00, + 0xb3, 0xc6, 0x00, + 0xc0, 0xc6, 0x02, + 0xc1, 0xc6, 0x00, + 0xc2, 0xc6, 0x00, + 0xc3, 0xc6, 0x00, + 0xd0, 0xc6, 0x03, + 0xd1, 0xc6, 0xfe, + 0xd2, 0xc6, 0x89, + 0xd3, 0xc6, 0xfa, + 0xe0, 0xc6, 0x1e, + 0xe1, 0xc6, 0x01, + 0xe2, 0xc6, 0x74, + 0xe3, 0xc6, 0xf6, + 0x40, 0xc5, 0x0d, + 0x30, 0xc7, 0x15, + 0x31, 0xc7, 0x7c, + 0x32, 0xc7, 0x0f, + 0x33, 0xc7, 0xa0, + 0x40, 0xc7, 0x00, + 0x41, 0xc7, 0x00, + 0x42, 0xc7, 0xf8, + 0x43, 0xc7, 0xf8, + 0x50, 0xc7, 0x00, + 0x51, 0xc7, 0x00, + 0x52, 0xc7, 0x00, + 0x53, 0xc7, 0x01, + 0x90, 0xc7, 0x00, + 0x91, 0xc7, 0x14, + 0x92, 0xc7, 0x00, + 0x93, 0xc7, 0x14, + 0xa0, 0xc7, 0x00, + 0xa1, 0xc7, 0x00, + 0xa2, 0xc7, 0xf8, + 0xa3, 0xc7, 0xf8, + 0xb0, 0xc7, 0x00, + 0xb1, 0xc7, 0x00, + 0xb2, 0xc7, 0x00, + 0xb3, 0xc7, 0x00, + 0x60, 0xc7, 0x03, + 0x61, 0xc7, 0xe8, + 0x62, 0xc7, 0x03, + 0x63, 0xc7, 0xb6, + 0x70, 0xc7, 0x00, + 0x71, 0xc7, 0x00, + 0x72, 0xc7, 0xf8, + 0x73, 0xc7, 0xf8, + 0x80, 0xc7, 0x00, + 0x81, 0xc7, 0x00, + 0x82, 0xc7, 0x00, + 0x83, 0xc7, 0x00, + 0xc0, 0xc7, 0x00, + 0xc1, 0xc7, 0x14, + 0xc2, 0xc7, 0x00, + 0xc3, 0xc7, 0x14, + 0xd0, 0xc7, 0x00, + 0xd1, 0xc7, 0x00, + 0xd2, 0xc7, 0xf8, + 0xd3, 0xc7, 0xf8, + 0xe0, 0xc7, 0x00, + 0xe1, 0xc7, 0x00, + 0xe2, 0xc7, 0x00, + 0xe3, 0xc7, 0x00, + 0x60, 0xc5, 0x02, + 0x61, 0xc5, 0x00, + 0x62, 0xc5, 0x00, + 0x63, 0xc5, 0x00, + 0x70, 0xc5, 0x02, + 0x71, 0xc5, 0x00, + 0x72, 0xc5, 0x00, + 0x73, 0xc5, 0x00, + 0x80, 0xc5, 0x02, + 0x81, 0xc5, 0x00, + 0x82, 0xc5, 0x00, + 0x83, 0xc5, 0x00, + 0x90, 0xc5, 0x02, + 0x91, 0xc5, 0x00, + 0x92, 0xc5, 0x00, + 0x93, 0xc5, 0x00, + 0x50, 0xc5, 0x01, +}; + +#endif diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index e2457738a332..772af23e7497 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -83,6 +83,7 @@ struct mc_private { bool idisp_codec; struct snd_soc_jack sdw_headset; struct device *headset_codec_dev; /* only one headset per card */ + struct device *amp_dev1, *amp_dev2; }; extern unsigned long sof_sdw_quirk; @@ -132,6 +133,7 @@ int sof_sdw_rt1308_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); +int sof_sdw_rt1308_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT1316 support */ int sof_sdw_rt1316_init(struct snd_soc_card *card, diff --git a/sound/soc/intel/boards/sof_sdw_rt1308.c b/sound/soc/intel/boards/sof_sdw_rt1308.c index f078fb1aad02..a19b055b9c6f 100644 --- a/sound/soc/intel/boards/sof_sdw_rt1308.c +++ b/sound/soc/intel/boards/sof_sdw_rt1308.c @@ -11,9 +11,86 @@ #include <sound/soc.h> #include <sound/soc-acpi.h> #include <sound/soc-dapm.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/dmi.h> #include "sof_sdw_common.h" +#include "sof_sdw_amp_coeff_tables.h" #include "../../codecs/rt1308.h" +struct rt1308_platform_data { + const unsigned char *bq_params; + const unsigned int bq_params_cnt; +}; + +static const struct rt1308_platform_data dell_0a5d_platform_data = { + .bq_params = dell_0a5d_bq_params, + .bq_params_cnt = ARRAY_SIZE(dell_0a5d_bq_params), +}; + +static const struct dmi_system_id dmi_platform_data[] = { + /* CometLake devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990") + }, + .driver_data = (void *)&dell_0a5d_platform_data, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F") + }, + .driver_data = (void *)&dell_0a5d_platform_data, + }, + /* TigerLake devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5D") + }, + .driver_data = (void *)&dell_0a5d_platform_data, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E") + }, + .driver_data = (void *)&dell_0a5d_platform_data, + }, +}; + +static int rt1308_add_device_props(struct device *sdw_dev) +{ + struct property_entry props[3] = {}; + struct fwnode_handle *fwnode; + const struct dmi_system_id *dmi_data; + const struct rt1308_platform_data *pdata; + unsigned char params[RT1308_MAX_BQ_REG]; + int ret; + + dmi_data = dmi_first_match(dmi_platform_data); + if (!dmi_data) + return 0; + + pdata = dmi_data->driver_data; + memcpy(&params, pdata->bq_params, sizeof(unsigned char) * pdata->bq_params_cnt); + + props[0] = PROPERTY_ENTRY_U8_ARRAY("realtek,bq-params", params); + props[1] = PROPERTY_ENTRY_U32("realtek,bq-params-cnt", pdata->bq_params_cnt); + + fwnode = fwnode_create_software_node(props, NULL); + if (IS_ERR(fwnode)) + return PTR_ERR(fwnode); + + ret = device_add_software_node(sdw_dev, to_software_node(fwnode)); + + fwnode_handle_put(fwnode); + + return ret; +} + static const struct snd_soc_dapm_widget rt1308_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), }; @@ -127,12 +204,33 @@ struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { .hw_params = rt1308_i2s_hw_params, }; +int sof_sdw_rt1308_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->amp_dev1) { + device_remove_software_node(ctx->amp_dev1); + put_device(ctx->amp_dev1); + } + + if (ctx->amp_dev2) { + device_remove_software_node(ctx->amp_dev2); + put_device(ctx->amp_dev2); + } + + return 0; +} + int sof_sdw_rt1308_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) { + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct device *sdw_dev1, *sdw_dev2; + int ret; + /* Count amp number and do init on playback link only. */ if (!playback) return 0; @@ -142,6 +240,28 @@ int sof_sdw_rt1308_init(struct snd_soc_card *card, dai_links->init = first_spk_init; if (info->amp_num == 2) { + sdw_dev1 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name); + if (!sdw_dev1) + return -EPROBE_DEFER; + + ret = rt1308_add_device_props(sdw_dev1); + if (ret < 0) { + put_device(sdw_dev1); + return ret; + } + ctx->amp_dev1 = sdw_dev1; + + sdw_dev2 = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[1].name); + if (!sdw_dev2) + return -EPROBE_DEFER; + + ret = rt1308_add_device_props(sdw_dev2); + if (ret < 0) { + put_device(sdw_dev2); + return ret; + } + ctx->amp_dev2 = sdw_dev2; + /* * if two 1308s are in one dai link, the init function * in this dai link will be first set for the first speaker, -- 2.38.1

2 1

[PATCH 2/4] ASoC: rt1316-sdw: get BQ params property and apply them
by shumingf＠realtek.com 11 Nov '22

11 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> If the machine driver level sets the BQ params into the device property, the codec driver will get the BQ params and apply them. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reviewed-by: Pierre-Louis Bossart pierre-louis.bossart(a)linux.intel.com --- sound/soc/codecs/rt1316-sdw.c | 39 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1316-sdw.h | 2 ++ 2 files changed, 41 insertions(+) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index ed0a11436362..2db7ee6c6d33 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -254,6 +254,17 @@ static int rt1316_read_prop(struct sdw_slave *slave) return 0; } +static void rt1316_apply_bq_params(struct rt1316_sdw_priv *rt1316) +{ + unsigned int i, reg, data; + + for (i = 0; i < rt1316->bq_params_cnt; i += 3) { + reg = rt1316->bq_params[i] | (rt1316->bq_params[i + 1] << 8); + data = rt1316->bq_params[i + 2]; + regmap_write(rt1316->regmap, reg, data); + } +} + static int rt1316_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(dev); @@ -590,14 +601,42 @@ static struct sdw_slave_ops rt1316_slave_ops = { .update_status = rt1316_update_status, }; +static int rt1316_sdw_parse_dt(struct rt1316_sdw_priv *rt1316, struct device *dev) +{ + int ret = 0; + + device_property_read_u32(dev, "realtek,bq-params-cnt", &rt1316->bq_params_cnt); + if (rt1316->bq_params_cnt) { + rt1316->bq_params = devm_kzalloc(dev, rt1316->bq_params_cnt, GFP_KERNEL); + if (!rt1316->bq_params) { + dev_err(dev, "Could not allocate bq_params memory\n"); + ret = -ENOMEM; + } else { + ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1316->bq_params, rt1316->bq_params_cnt); + if (ret < 0) + dev_err(dev, "Could not read list of realtek,bq-params\n"); + } + } + + dev_dbg(dev, "bq_params_cnt=%d\n", rt1316->bq_params_cnt); + return ret; +} + static int rt1316_sdw_component_probe(struct snd_soc_component *component) { + struct rt1316_sdw_priv *rt1316 = snd_soc_component_get_drvdata(component); int ret; + rt1316->component = component; + rt1316_sdw_parse_dt(rt1316, &rt1316->sdw_slave->dev); + ret = pm_runtime_resume(component->dev); if (ret < 0 && ret != -EACCES) return ret; + /* apply BQ params */ + rt1316_apply_bq_params(rt1316); + return 0; } diff --git a/sound/soc/codecs/rt1316-sdw.h b/sound/soc/codecs/rt1316-sdw.h index cbcdaa8f8cfa..57dbd49993b0 100644 --- a/sound/soc/codecs/rt1316-sdw.h +++ b/sound/soc/codecs/rt1316-sdw.h @@ -46,6 +46,8 @@ struct rt1316_sdw_priv { struct sdw_bus_params params; bool hw_init; bool first_hw_init; + unsigned char *bq_params; + unsigned int bq_params_cnt; }; struct sdw_stream_data { -- 2.38.1

2 1

[PATCH 1/4] ASoC: rt1308-sdw: get BQ params property and apply them
by shumingf＠realtek.com 11 Nov '22

11 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> If the machine driver level sets the BQ params into the device property, the codec driver will get the BQ params and apply them. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reviewed-by: Pierre-Louis Bossart pierre-louis.bossart(a)linux.intel.com --- sound/soc/codecs/rt1308-sdw.c | 39 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1308-sdw.h | 2 ++ 2 files changed, 41 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index f99aed353f10..7f4248284f35 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -197,6 +197,17 @@ static void rt1308_apply_calib_params(struct rt1308_sdw_priv *rt1308) efuse_c_btl_l, efuse_c_btl_r); } +static void rt1308_apply_bq_params(struct rt1308_sdw_priv *rt1308) +{ + unsigned int i, reg, data; + + for (i = 0; i < rt1308->bq_params_cnt; i += 3) { + reg = rt1308->bq_params[i] | (rt1308->bq_params[i + 1] << 8); + data = rt1308->bq_params[i + 2]; + regmap_write(rt1308->regmap, reg, data); + } +} + static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); @@ -619,14 +630,42 @@ static const struct sdw_slave_ops rt1308_slave_ops = { .bus_config = rt1308_bus_config, }; +static int rt1308_sdw_parse_dt(struct rt1308_sdw_priv *rt1308, struct device *dev) +{ + int ret = 0; + + device_property_read_u32(dev, "realtek,bq-params-cnt", &rt1308->bq_params_cnt); + if (rt1308->bq_params_cnt) { + rt1308->bq_params = devm_kzalloc(dev, rt1308->bq_params_cnt, GFP_KERNEL); + if (!rt1308->bq_params) { + dev_err(dev, "Could not allocate bq_params memory\n"); + ret = -ENOMEM; + } else { + ret = device_property_read_u8_array(dev, "realtek,bq-params", rt1308->bq_params, rt1308->bq_params_cnt); + if (ret < 0) + dev_err(dev, "Could not read list of realtek,bq-params\n"); + } + } + + dev_dbg(dev, "bq_params_cnt=%d\n", rt1308->bq_params_cnt); + return ret; +} + static int rt1308_sdw_component_probe(struct snd_soc_component *component) { + struct rt1308_sdw_priv *rt1308 = snd_soc_component_get_drvdata(component); int ret; + rt1308->component = component; + rt1308_sdw_parse_dt(rt1308, &rt1308->sdw_slave->dev); + ret = pm_runtime_resume(component->dev); if (ret < 0 && ret != -EACCES) return ret; + /* apply BQ params */ + rt1308_apply_bq_params(rt1308); + return 0; } diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 62ce27799307..1eaaef9f351b 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -166,6 +166,8 @@ struct rt1308_sdw_priv { int rx_mask; int slots; int hw_ver; + unsigned char *bq_params; + unsigned int bq_params_cnt; }; struct sdw_stream_data { -- 2.38.1

2 1

[PATCH] ASoC: dt-bindings: qcom,wsa883x: Add sound-name-prefix
by Krzysztof Kozlowski 11 Nov '22

11 Nov '22

For multiple speakers attached, it's useful to give them some name. Sound core already supports this, so reference name-prefix.yaml. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml index 99f9c10bbc83..f3fb1de93ba2 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -15,6 +15,9 @@ description: | Their primary operating mode uses a SoundWire digital audio interface. This binding is for SoundWire interface. +allOf: + - $ref: name-prefix.yaml# + properties: compatible: const: sdw10217020200 @@ -35,6 +38,8 @@ properties: '#sound-dai-cells': const: 0 + sound-name-prefix: true + required: - compatible - reg @@ -61,6 +66,7 @@ examples: vdd-supply = <&vreg_s10b_1p8>; #thermal-sensor-cells = <0>; #sound-dai-cells = <0>; + sound-name-prefix = "SpkrLeft"; }; speaker@0,2 { @@ -70,6 +76,7 @@ examples: vdd-supply = <&vreg_s10b_1p8>; #thermal-sensor-cells = <0>; #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; }; }; -- 2.34.1

2 3

[PATCH 1/2] ASoC: dt-bindings: qcom, lpass: do not hard-code clock-output-names
by Krzysztof Kozlowski 11 Nov '22

11 Nov '22

The purpose of clock-output-names is to customize desired clock name, not use one, same name. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/qcom,lpass-rx-macro.yaml | 3 +-- .../devicetree/bindings/sound/qcom,lpass-tx-macro.yaml | 3 +-- .../devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 3 +-- .../devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml | 3 +-- 4 files changed, 4 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index 1de11e7f33bb..14016671f32b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -43,8 +43,7 @@ properties: - const: fsgen clock-output-names: - items: - - const: mclk + maxItems: 1 power-domains: maxItems: 2 diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml index de8297b358e8..e647ba392a0f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -43,8 +43,7 @@ properties: - const: fsgen clock-output-names: - items: - - const: mclk + maxItems: 1 power-domains: maxItems: 2 diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index 9f473c08cb2e..c36caf90b837 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -39,8 +39,7 @@ properties: - const: mclk clock-output-names: - items: - - const: fsgen + maxItems: 1 power-domains: maxItems: 2 diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 4959ad658eac..155c7344412a 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -38,8 +38,7 @@ properties: - const: fsgen clock-output-names: - items: - - const: mclk + maxItems: 1 qcom,dmic-sample-rate: description: dmic sample rate -- 2.34.1

3 5

missing sound on kernel-5.15
by chihhao chen 11 Nov '22

11 Nov '22

I am testing many headsets on Android platform with different kernel versions. On kernel-5.10, calling sequence for playback is 1. snd_usb_hw_params -> set highest sampling rate, no SAMPLING_FREQ_CONTROL USB request sent in this stage 2. snd_pcm_release 3. snd_usb_hw_params -> set proper sampling rate, still no SAMPLING_FREQ_CONTROL USB request 4. snd_usb_pcm_parepare -> in configure_endpoint SAMPLING_FREQ_CONTROL USB request was sent on USB bus With the same calling sequence, a slightly behavior change on kernel- 5.15 is 1. snd_usb_hw_params -> set highest sampling rate, SAMPLING_FREQ_CONTROL USB request was sent on USB bus 2. snd_pcm_release 3. snd_usb_hw_params -> set proper sampling rate, SAMPLING_FREQ_CONTROL USB request was sent too (because configure_endpoint was moved to snd_usb_hw_params function) 4. snd_usb_pcm_parepare -> no SAMPLING_FREQ_CONTROL USB request because of USB EP flag check I checked all USB packets and confirmed audio data output was correct. But there may be missing sound problem for the first sound with many headsets (for example Samsung headset). I found this issue is related to two-times sampling rate set request. (I tried to forcibly skip first USB request, everything became okay.) So do you guys know why configure_endpoint was moved to snd_usb_hw_params? Or could you provide patch for this problem? Many thanks Chihhao

4 26

[PATCH] soundwire: intel: Initialize clock stop timeout
by Bard Liao 11 Nov '22

11 Nov '22

From: Sjoerd Simons <sjoerd(a)collabora.com> The bus->clk_stop_timeout member is only initialized to a non-zero value during the codec driver probe. This can lead to corner cases where this value remains pegged at zero when the bus suspends, which results in an endless loop in sdw_bus_wait_for_clk_prep_deprep(). Corner cases include configurations with no codecs described in the firmware, or delays in probing codec drivers. Initializing the default timeout to the smallest non-zero value avoid this problem and allows for the existing logic to be preserved: the bus->clk_stop_timeout is set as the maximum required by all codecs connected on the bus. Fixes: 1f2dcf3a154ac ("soundwire: intel: set dev_num_ida_min") Signed-off-by: Sjoerd Simons <sjoerd(a)collabora.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 25ec9c272239..78d35bb4852c 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1311,6 +1311,7 @@ static int intel_link_probe(struct auxiliary_device *auxdev, bus->link_id = auxdev->id; bus->dev_num_ida_min = INTEL_DEV_NUM_IDA_MIN; + bus->clk_stop_timeout = 1; sdw_cdns_probe(cdns); -- 2.25.1

3 4

[PATCH] ASoC: tegra: Fix spelling mistake "fliter" -> "filter"
by Colin Ian King 10 Nov '22

10 Nov '22

There is a spelling mistake in struct member fliter_structure. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/tegra/tegra210_mbdrc.c | 4 ++-- sound/soc/tegra/tegra210_mbdrc.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra210_mbdrc.c b/sound/soc/tegra/tegra210_mbdrc.c index d786daa6aba6..eeacb1220268 100644 --- a/sound/soc/tegra/tegra210_mbdrc.c +++ b/sound/soc/tegra/tegra210_mbdrc.c @@ -54,7 +54,7 @@ static const struct tegra210_mbdrc_config mbdrc_init_config = { .mode = 0, /* Bypass */ .rms_off = 48, .peak_rms_mode = 1, /* PEAK */ - .fliter_structure = 0, /* All-pass tree */ + .filter_structure = 0, /* All-pass tree */ .shift_ctrl = 30, .frame_size = 32, .channel_mask = 0x3, @@ -819,7 +819,7 @@ int tegra210_mbdrc_component_init(struct snd_soc_component *cmpnt) regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_MASK, - conf->fliter_structure << + conf->filter_structure << TEGRA210_MBDRC_CFG_FILTER_STRUCTURE_SHIFT); regmap_update_bits(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, diff --git a/sound/soc/tegra/tegra210_mbdrc.h b/sound/soc/tegra/tegra210_mbdrc.h index 4c48da0e1dea..76e443a7934e 100644 --- a/sound/soc/tegra/tegra210_mbdrc.h +++ b/sound/soc/tegra/tegra210_mbdrc.h @@ -199,7 +199,7 @@ struct tegra210_mbdrc_config { unsigned int mode; unsigned int rms_off; unsigned int peak_rms_mode; - unsigned int fliter_structure; + unsigned int filter_structure; unsigned int shift_ctrl; unsigned int frame_size; unsigned int channel_mask; -- 2.38.1

3 2

[PATCH] ASoC: sgtl5000: Reset the CHIP_CLK_CTRL reg on remove
by Detlev Casanova 10 Nov '22

10 Nov '22

Since commit bf2aebccddef ("ASoC: sgtl5000: Fix noise on shutdown/remove"), the device power control registers are reset when the driver is removed/shutdown. This is an issue when the device is configured to use the PLL clock. The device will stop responding if it is still configured to use the PLL clock but the PLL clock is powered down. When rebooting linux, the probe function will show: sgtl5000 0-000a: Error reading chip id -11 Make sure that the CHIP_CLK_CTRL is reset to its default value before powering down the device. Fixes: bf2aebccddef ("ASoC: sgtl5000: Fix noise on shutdown/remove") Signed-off-by: Detlev Casanova <detlev.casanova(a)collabora.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4b2135eba74d..a916f4619ea3 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1794,6 +1794,7 @@ static void sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_CLK_CTRL, SGTL5000_CHIP_CLK_CTRL_DEFAULT); regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); -- 2.38.1

3 2

[PATCH] ASoC: stm32: i2s: remove irqf_oneshot flag
by Olivier Moysan 10 Nov '22

10 Nov '22

The IRQF_ONESHOT flag allows to ensure that the interrupt is not unmasked after the hard interrupt context handler has been executed and the thread has been woken. The interrupt line is unmasked after the thread handler function has been executed. The STM32 I2S driver does not implement a threaded IRQ handler. So, the IRQF_ONESHOT flag is not useful in I2S driver. Remove this flag to allow the interrupt routine to be managed as a thread in RT mode. Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> --- sound/soc/stm/stm32_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index ce7f6942308f..f3dd9f8e621c 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -1077,7 +1077,7 @@ static int stm32_i2s_parse_dt(struct platform_device *pdev, if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, IRQF_ONESHOT, + ret = devm_request_irq(&pdev->dev, irq, stm32_i2s_isr, 0, dev_name(&pdev->dev), i2s); if (ret) { dev_err(&pdev->dev, "irq request returned %d\n", ret); -- 2.25.1

2 1

[PATCH] ASoC: hdac_hda: fix hda pcm buffer overflow issue
by Junxiao Chang 10 Nov '22

10 Nov '22

When KASAN is enabled, below log might be dumped with Intel EHL hardware: [ 48.583597] ================================================================== [ 48.585921] BUG: KASAN: slab-out-of-bounds in hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.587995] Write of size 4 at addr ffff888103489708 by task pulseaudio/759 [ 48.589237] CPU: 2 PID: 759 Comm: pulseaudio Tainted: G U E 5.15.71-intel-ese-standard-lts #9 [ 48.591272] Hardware name: Intel Corporation Elkhart Lake Embedded Platform/ElkhartLake LPDDR4x T3 CRB, BIOS EHLSFWI1.R00.4251.A01.2206130432 06/13/2022 [ 48.593010] Call Trace: [ 48.593648] <TASK> [ 48.593852] dump_stack_lvl+0x34/0x48 [ 48.594404] print_address_description.constprop.0+0x1f/0x140 [ 48.595174] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.595868] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.596519] kasan_report.cold+0x7f/0x11b [ 48.597003] ? hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] [ 48.597885] hdac_hda_dai_hw_params+0x20a/0x22b [snd_soc_hdac_hda] HDAC_LAST_DAI_ID is last index id, pcm buffer array size should be +1 to avoid out of bound access. Fixes: 608b8c36c371 ("ASoC: hdac_hda: add support for HDMI/DP as a HDA codec") Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Junxiao Chang <junxiao.chang(a)intel.com> Signed-off-by: Furong Zhou <furong.zhou(a)intel.com> --- sound/soc/codecs/hdac_hda.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index fc19c34ca00e5..b65560981abb2 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -14,7 +14,7 @@ enum { HDAC_HDMI_1_DAI_ID, HDAC_HDMI_2_DAI_ID, HDAC_HDMI_3_DAI_ID, - HDAC_LAST_DAI_ID = HDAC_HDMI_3_DAI_ID, + HDAC_DAI_ID_NUM }; struct hdac_hda_pcm { @@ -24,7 +24,7 @@ struct hdac_hda_pcm { struct hdac_hda_priv { struct hda_codec *codec; - struct hdac_hda_pcm pcm[HDAC_LAST_DAI_ID]; + struct hdac_hda_pcm pcm[HDAC_DAI_ID_NUM]; bool need_display_power; }; -- 2.25.1

2 1

[PATCH v2] ASoC: wm8962: Wait for updated value of WM8962_CLOCKING1 register
by Chancel Liu 10 Nov '22

10 Nov '22

DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate correct frequency of LRCLK and BCLK. Sometimes the read-only value can't be updated timely after enabling SYSCLK. This results in wrong calculation values. Delay is introduced here to wait for newest value from register. The time of the delay should be at least 500~1000us according to test. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b4b4355c6728..b901e4c65e8a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2503,6 +2503,14 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) snd_soc_component_update_bits(component, WM8962_CLOCKING2, WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + /* DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate + * correct frequency of LRCLK and BCLK. Sometimes the read-only value + * can't be updated timely after enabling SYSCLK. This results in wrong + * calculation values. Delay is introduced here to wait for newest + * value from register. The time of the delay should be at least + * 500~1000us according to test. + */ + usleep_range(500, 1000); dspclk = snd_soc_component_read(component, WM8962_CLOCKING1); if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_ON) -- 2.25.1

3 2

[PATCH 0/9] ASoC: mediatek: Register machine drivers to module device table
by Nícolas F. R. A. Prado 10 Nov '22

10 Nov '22

Some ASoC machine drivers for MediaTek platforms were not being registered to the module device table, preventing them to be automatically loaded when built as modules. This series adds the missing registration to the drivers. Nícolas F. R. A. Prado (9): ASoC: mediatek: mt2701-cs42448: Register to module device table ASoC: mediatek: mt2701-wm8960: Register to module device table ASoC: mediatek: mt6797-mt6351: Register to module device table ASoC: mediatek: mt8183-da7219: Register to module device table ASoC: mediatek: mt8183-mt6358: Register to module device table ASoC: mediatek: mt8186-da7219: Register to module device table ASoC: mediatek: mt8186-rt5682: Register to module device table ASoC: mediatek: mt8192-mt6359: Register to module device table ASoC: mediatek: mt8195-mt6359: Register to module device table sound/soc/mediatek/mt2701/mt2701-cs42448.c | 1 + sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 + sound/soc/mediatek/mt6797/mt6797-mt6351.c | 1 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 + sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 1 + sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 1 + sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 1 + sound/soc/mediatek/mt8195/mt8195-mt6359.c | 1 + 9 files changed, 9 insertions(+) -- 2.38.1

4 21

[PATCH 0/3] ASoC: SOF: get pipeline instance id from pipe_widget->instance_id
by Bard Liao 10 Nov '22

10 Nov '22

Currently we set pipeline instance id = swidget->pipeline_id, but pipeline_id is from topology and can be any number. In fact, the number of pipelines is limited. This patch use ida to allocate pipeline instance id and will be used for pipeline instance id in the IPC message. Bard Liao (3): ASoC: SOF: ipc4-topology: rename sof_ipc4_widget_free_comp ASoC: SOF: ipc4-loader: get max pipeline number ASoC: SOF: ipc4: get pipeline instance id from pipe_widget->instance_id sound/soc/sof/intel/hda-dai.c | 12 ++++++------ sound/soc/sof/ipc4-loader.c | 11 +++++++++++ sound/soc/sof/ipc4-pcm.c | 4 ++-- sound/soc/sof/ipc4-priv.h | 2 ++ sound/soc/sof/ipc4-topology.c | 30 ++++++++++++++++++++++++------ 5 files changed, 45 insertions(+), 14 deletions(-) -- 2.25.1

2 4

[PATCH 0/3] ASoC: SOF: Intel: update D0i3 registers for MTL
by Pierre-Louis Bossart 10 Nov '22

10 Nov '22

MeteorLake relies on a different register for D0i3 configuration, add a platform-specific callback to abstract the differences. Rander Wang (3): ASoC: SOF: Intel: add d0i3 definition for MTL ASoC: SOF: Intel: add d0i3_offset in chip_info ASoC: SOF: Intel: set d0i3 register with d0i3_offset sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 2 ++ sound/soc/sof/intel/hda-dsp.c | 21 ++++++++++++++------- sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/mtl.c | 1 + sound/soc/sof/intel/mtl.h | 2 ++ sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/tgl.c | 4 ++++ 8 files changed, 26 insertions(+), 7 deletions(-) -- 2.34.1

2 4

[PATCH 0/3] ASoC: SOF: ipc4-topology: Add widget queue support
by Peter Ujfalusi 10 Nov '22

10 Nov '22

Hi, with SOF topology2 for IPC4, widgets might have mutliple queues they can be connected. The queues to use between components are descibed in the topology file. This series adds widget queue support (specify which pin to connect) for ipc4-topology with topology2. Note: currently queue 0 of a widget is used as hardwired default. Regards, Peter --- Chao Song (3): ASoC: SOF: Add support for parsing the number of sink/source pins ASoC: SOF: Add support to parse pin binding array from topology ASoC: SOF: topology: Add helper to get/put widget queue id include/uapi/sound/sof/tokens.h | 9 ++ sound/soc/sof/ipc4-topology.c | 133 +++++++++++++++++++++--- sound/soc/sof/sof-audio.h | 42 ++++++++ sound/soc/sof/topology.c | 177 ++++++++++++++++++++++++++++++-- 4 files changed, 340 insertions(+), 21 deletions(-) -- 2.38.1

2 4

[PATCH] ASoC: SOF: ipc4-topology: Implement tear_down_all_pipelines callback
by Peter Ujfalusi 10 Nov '22

10 Nov '22

During system suspend there is a chance that the running stream undergo an xrun and instead of the expected SNDRV_PCM_TRIGGER_SUSPEND trigger we will receive SNDRV_PCM_TRIGGER_STOP. The handling of SUSPEND and STOP triggers differ that case of the later the sof_pcm_stream_free() will be called with free_widget_list = false. But we must make sure that all active widgets and streams are free before entering suspend in order to be able to resume without error. We can utilize the tear_down_all_pipelines to put the system to an expected state and to fix the suspend/resume error. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index ab85dde4303b..c31b7f065c1d 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1813,6 +1813,39 @@ static int sof_ipc4_dai_get_clk(struct snd_sof_dev *sdev, struct snd_sof_dai *da return -EINVAL; } +static int sof_ipc4_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verify) +{ + struct snd_sof_pcm *spcm; + int dir, ret; + + /* + * This function is called during system suspend, we need to make sure + * that all streams have been freed up. + * Freeing might have been skipped when xrun happened just at the start + * of the suspend and it sent a SNDRV_PCM_TRIGGER_STOP to the active + * stream. This will call sof_pcm_stream_free() with + * free_widget_list = false which will leave the kernel and firmware out + * of sync during suspend/resume. + * + * This will also make sure that paused streams handled correctly. + */ + list_for_each_entry(spcm, &sdev->pcm_list, list) { + for_each_pcm_streams(dir) { + struct snd_pcm_substream *substream = spcm->stream[dir].substream; + + if (!substream || !substream->runtime) + continue; + + if (spcm->stream[dir].list) { + ret = sof_pcm_stream_free(sdev, substream, spcm, dir, true); + if (ret < 0) + return ret; + } + } + } + return 0; +} + static enum sof_tokens host_token_list[] = { SOF_COMP_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, @@ -1913,4 +1946,5 @@ const struct sof_ipc_tplg_ops ipc4_tplg_ops = { .dai_config = sof_ipc4_dai_config, .parse_manifest = sof_ipc4_parse_manifest, .dai_get_clk = sof_ipc4_dai_get_clk, + .tear_down_all_pipelines = sof_ipc4_tear_down_all_pipelines, }; -- 2.38.1

2 1

[PATCH] ASoC: SOF: Intel: hda-loader: use small buffer for iccmax stream
by Peter Ujfalusi 10 Nov '22

10 Nov '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> The data received via iccmax stream is not used anywhere, so no need to allocate a big DMA buffer for it. This is especially important as the allocation is done even in cases where reload of the firmware is skipped and execution happens directly from the firmware stored in IMR. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 3a4b0b6e2c5c..d680562edb35 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -322,7 +322,6 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) { struct hdac_ext_stream *iccmax_stream; struct hdac_bus *bus = sof_to_bus(sdev); - struct firmware stripped_firmware; struct snd_dma_buffer dmab_bdl; int ret, ret1; u8 original_gb; @@ -330,15 +329,11 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) /* save the original LTRP guardband value */ original_gb = snd_hdac_chip_readb(bus, VS_LTRP) & HDA_VS_INTEL_LTRP_GB_MASK; - if (sdev->basefw.fw->size <= sdev->basefw.payload_offset) { - dev_err(sdev->dev, "error: firmware size must be greater than firmware offset\n"); - return -EINVAL; - } - - stripped_firmware.size = sdev->basefw.fw->size - sdev->basefw.payload_offset; - - /* prepare capture stream for ICCMAX */ - iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, + /* + * Prepare capture stream for ICCMAX. We do not need to store + * the data, so use a buffer of PAGE_SIZE for receiving. + */ + iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, PAGE_SIZE, &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE); if (IS_ERR(iccmax_stream)) { dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n"); -- 2.38.1

2 1

[PATCH 1/2] ASoC: mediatek: mt8183-da7219: Register to module device table
by Nícolas F. R. A. Prado 10 Nov '22

10 Nov '22

Register the compatibles for this module on the module device table so it can be automatically loaded when a matching device is found on the system. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 9f22d3939818..0e572fe28c58 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -842,6 +842,7 @@ static const struct of_device_id mt8183_da7219_max98357_dt_match[] = { }, {} }; +MODULE_DEVICE_TABLE(of, mt8183_da7219_max98357_dt_match); #endif static struct platform_driver mt8183_da7219_max98357_driver = { -- 2.38.1

3 5

[PATCH 0/3] ASoC: simpe-card/audio-graph fixups
by Kuninori Morimoto 10 Nov '22

10 Nov '22

Hi Mark These are simpe-card/audio-graph fixup patches. Kuninori Morimoto (3): ASoC: simple-card-utils: remove asoc_simple_convert_fixup() ASoC: audio-graph-card2-custom-sample.dtsi: add convert-rate for DPCM (MIXer) ASoC: audio-graph-card2: check also dpcm node for convert-xxx include/sound/simple_card_utils.h | 2 -- .../audio-graph-card2-custom-sample.dtsi | 10 ++++-- sound/soc/generic/audio-graph-card2.c | 3 +- sound/soc/generic/simple-card-utils.c | 35 +++++++------------ 4 files changed, 23 insertions(+), 27 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

2 4

[PATCH 0/7] ASoC: codecs: da7219: Do not export da7219_aad_jack_det()
by Cezary Rojewski 10 Nov '22

10 Nov '22

Goal of the series is to drop any usage of the internal function. First, update all users to utilize component->set_jack() and then remove the relevant EXPORT_SYMBOL_GPL. Amadeusz Sławiński (2): ASoC: codecs: da7219: Introduce set_jack() callback ASoC: codecs: da7219: Do not export internal symbols Cezary Rojewski (5): ASoC: Intel: Drop da7219_aad_jack_det() usage ASoC: mediatek: Drop da7219_aad_jack_det() usage ASoC: rockchip: Drop da7219_aad_jack_det() usage ASoC: amd: Drop da7219_aad_jack_det() usage ASoC: Intel: avs: Drop da7219_aad_jack_det() usage sound/soc/amd/acp-da7219-max98357a.c | 3 +-- sound/soc/codecs/da7219-aad.c | 4 ---- sound/soc/codecs/da7219.c | 9 +++++++++ sound/soc/intel/avs/boards/da7219.c | 3 +-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 3 +-- sound/soc/intel/boards/kbl_da7219_max98357a.c | 3 +-- sound/soc/intel/boards/kbl_da7219_max98927.c | 3 +-- sound/soc/intel/boards/sof_da7219_max98373.c | 4 ++-- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 3 +-- .../soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 4 ++-- sound/soc/rockchip/rk3399_gru_sound.c | 3 +-- 11 files changed, 20 insertions(+), 22 deletions(-) -- 2.25.1

2 8

[PATCH linux-next v2] ASoC: SOF: remove duplicated included sof-audio.h
by yang.yang29＠zte.com.cn 10 Nov '22

10 Nov '22

From: Xu Panda <xu.panda(a)zte.com.cn> The sof-audio.h is included more than once. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Xu Panda <xu.panda(a)zte.com.cn> Signed-off-by: Yang Yang <yang.yang29(a)zte.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- change for v2 - add maintainers and the alsa-devel mailing list in CC. --- sound/soc/sof/amd/acp-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 27b95187356e..348e70dfe2a5 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -13,7 +13,6 @@ #include "../sof-priv.h" #include "../sof-audio.h" #include "../ops.h" -#include "../sof-audio.h" #include "acp.h" #include "acp-dsp-offset.h" -- 2.15.2

3 2

[PATCH 0/2] ASoC: Intel: avs: DSP recovery and resume fixes
by Cezary Rojewski 10 Nov '22

10 Nov '22

Two fixes that are result of the recent discussions [1][2]. First adds missing locking around snd_pcm_stop() while the second fix sets substream state to DISCONNECTED if any suspend/resume related operation fails so that userspace has means to be aware that something went wrong during said operation. [1]: https://lore.kernel.org/alsa-devel/73e6425f-8e51-e26f-ad83-ccc5df517a43@int… [2]: https://lore.kernel.org/alsa-devel/20221104131244.3920179-1-cezary.rojewski… Cezary Rojewski (2): ASoC: Intel: avs: Lock substream before snd_pcm_stop() ASoC: Intel: avs: Disconnect substream if suspend or resume fails sound/soc/intel/avs/ipc.c | 2 ++ sound/soc/intel/avs/pcm.c | 10 ++++++++-- 2 files changed, 10 insertions(+), 2 deletions(-) -- 2.25.1

2 8

[PATCH] ALSA: hda: fix potential memleak in 'add_widget_node'
by Ye Bin 10 Nov '22

10 Nov '22

From: Ye Bin <yebin10(a)huawei.com> As 'kobject_add' may allocated memory for 'kobject->name' when return error. And in this function, if call 'kobject_add' failed didn't free kobject. So call 'kobject_put' to recycling resources. Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/hda/hdac_sysfs.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index e47de49a32e3..62a9615dcf52 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -346,8 +346,10 @@ static int add_widget_node(struct kobject *parent, hda_nid_t nid, return -ENOMEM; kobject_init(kobj, &widget_ktype); err = kobject_add(kobj, parent, "%02x", nid); - if (err < 0) + if (err < 0) { + kobject_put(kobj); return err; + } err = sysfs_create_group(kobj, group); if (err < 0) { kobject_put(kobj); -- 2.31.1

2 1

[PATCH] ASoC: codecs: wsa883x: Use proper shutdown GPIO values
by Krzysztof Kozlowski 10 Nov '22

10 Nov '22

The shutdown GPIO is active_low (sd_n), but this depends on actual board layout. Linux drivers should only care about logical state, where high (1) means shutdown and low (0) means do not shutdown. Invert the GPIO to match logical value. Fixes: 43b8c7dc85a1 ("ASoC: codecs: add wsa883x amplifier support") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/wsa883x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index c7b10bbfba7e..77a7dd3cf495 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -1392,7 +1392,7 @@ static int wsa883x_probe(struct sdw_slave *pdev, } wsa883x->sd_n = devm_gpiod_get_optional(&pdev->dev, "powerdown", - GPIOD_FLAGS_BIT_NONEXCLUSIVE); + GPIOD_FLAGS_BIT_NONEXCLUSIVE | GPIOD_OUT_HIGH); if (IS_ERR(wsa883x->sd_n)) { dev_err(&pdev->dev, "Shutdown Control GPIO not found\n"); ret = PTR_ERR(wsa883x->sd_n); @@ -1411,7 +1411,7 @@ static int wsa883x_probe(struct sdw_slave *pdev, pdev->prop.simple_clk_stop_capable = true; pdev->prop.sink_dpn_prop = wsa_sink_dpn_prop; pdev->prop.scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; - gpiod_direction_output(wsa883x->sd_n, 1); + gpiod_direction_output(wsa883x->sd_n, 0); wsa883x->regmap = devm_regmap_init_sdw(pdev, &wsa883x_regmap_config); if (IS_ERR(wsa883x->regmap)) { -- 2.34.1

1 1

[PATCH] ALSA: memalloc: Don't fall back for SG-buffer with IOMMU
by Takashi Iwai 10 Nov '22

10 Nov '22

When the non-contiguous page allocation for SG buffer allocation fails, the memalloc helper tries to fall back to the old page allocation methods. This would, however, result in the bogus page addresses when IOMMU is enabled. Usually in such a case, the fallback allocation should fail as well, but occasionally it succeeds and hitting a bad access. The fallback was thought for non-IOMMU case, and as the error from dma_alloc_noncontiguous() with IOMMU essentially implies a fatal memory allocation error, we should return the error straightforwardly without fallback. This avoids the corner case like the above. The patch also renames the local variable "dma_ops" with snd_ prefix for avoiding the name conflict. Fixes: a8d302a0b770 ("ALSA: memalloc: Revive x86-specific WC page allocations again") Reported-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/alpine.DEB.2.22.394.2211041541090.3532114@elitele… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 03cffe771366..6a81aaab25ab 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -9,6 +9,7 @@ #include <linux/slab.h> #include <linux/mm.h> #include <linux/dma-mapping.h> +#include <linux/dma-map-ops.h> #include <linux/genalloc.h> #include <linux/highmem.h> #include <linux/vmalloc.h> @@ -541,19 +542,20 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) struct sg_table *sgt; void *p; - sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, - DEFAULT_GFP, 0); - if (!sgt) { #ifdef CONFIG_SND_DMA_SGBUF + if (!get_dma_ops(dmab->dev.dev)) { if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; else dmab->dev.type = SNDRV_DMA_TYPE_DEV_SG_FALLBACK; return snd_dma_sg_fallback_alloc(dmab, size); -#else - return NULL; -#endif } +#endif + + sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, + DEFAULT_GFP, 0); + if (!sgt) + return NULL; dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, sg_dma_address(sgt->sgl)); @@ -857,7 +859,7 @@ static const struct snd_malloc_ops snd_dma_noncoherent_ops = { /* * Entry points */ -static const struct snd_malloc_ops *dma_ops[] = { +static const struct snd_malloc_ops *snd_dma_ops[] = { [SNDRV_DMA_TYPE_CONTINUOUS] = &snd_dma_continuous_ops, [SNDRV_DMA_TYPE_VMALLOC] = &snd_dma_vmalloc_ops, #ifdef CONFIG_HAS_DMA @@ -883,7 +885,7 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) if (WARN_ON_ONCE(!dmab)) return NULL; if (WARN_ON_ONCE(dmab->dev.type <= SNDRV_DMA_TYPE_UNKNOWN || - dmab->dev.type >= ARRAY_SIZE(dma_ops))) + dmab->dev.type >= ARRAY_SIZE(snd_dma_ops))) return NULL; - return dma_ops[dmab->dev.type]; + return snd_dma_ops[dmab->dev.type]; } -- 2.35.3

1 0

Re: regression with SG DMA buf allocations and IOMMU in low-mem
by Kai Vehmanen 10 Nov '22

10 Nov '22

Hi, [ and sorry, I had the bounces address in cc originally, routing reply to list as well ] On Fri, 4 Nov 2022, Takashi Iwai wrote: > On Fri, 04 Nov 2022 15:56:50 +0100, Kai Vehmanen wrote: > > I've been debugging a SOF DSP load fail on VT-D/IOMMU systems [1] and > > that brought me to these two commits from you: > > .. but in rare low-memory cases, the dma_alloc_noncontiguous() > > call will fail in core/memalloc.c and we go to the fallback path. > > > > So we go to snd_dma_sg_fallback_alloc(), but here it seems something is > > missing. The pages are not allocated with DMA mapping API anymore, so > > IOMMU won't know about the memory and in our case the DSP load will fail > > to a IOMMU fault. [...] > > Hrm, that's a tough issue. Basically if dma_alloc_noncontiguous() > fails, it's really out of memory -- at least under the situation where > IOMMU is enabled. The fallback path was introduced for the case where > there is no IOMMU and noncontiguous allocation fails from the > beginning. ack. This matches with the bug reports. These happen rarely and on systems with a lot of memory pressure. We have recently submitted a few features&fixes that will further reduce these allocs, so it probably is better outcome to have a plain out of memory error. > And, what makes things more complicated is that snd_dma_dev_alloc() > cannot be used for SG pages when IOMMU is involved. We can't get the > proper pages for mapping SG from the DMA address in that case; some > systems may work with virt_to_page() but it's not guranteed to work. Aa, ok, so we need to use dma_alloc_noncontiguous() to do this properly and if it returns ENOMEM, that's what it is then, right. > So, I believe there are two issues: > 1. The firmware allocation fails with dma_alloc_noncontiguous() with > IOMMU enabled > > 2. The helper goes to the fallback path and occasionally it worked, > although the pages can't be used with IOMMU. > > Basically when 1 happens, we should just return an error without going > to fallback path. But it won't "fix" your case? I think an explicit error would be best. The problem now is that the driver will think the allocation (and mapping to device) is fine and proceeds to program the hardware to use the address. This will then create an IOMMU fault down the line that is not so straighforward to recover from (worst case is that a full device level reset needs to be done). And given driver doesn't know it got a faulty mapping, it's hard to make the decision why the fault happened. Br, Kai

2 3

[PATCH] ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue
by Ai Chao 10 Nov '22

10 Nov '22

For Hamedal C20, the current rate is different from the runtime rate, snd_usb_endpoint stop and close endpoint to resetting rate. if snd_usb_endpoint close the endpoint, sometimes usb will disconnect the device. Signed-off-by: Ai Chao <aichao(a)kylinos.cn> --- sound/usb/endpoint.c | 3 ++- sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 3 +++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d0b8d61d1d22..310cd6fb0038 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -931,7 +931,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened) + if (!--ep->iface_ref->opened && + !(chip->quirk_flags & QUIRK_FLAG_IFACE_SKIP_CLOSE)) endpoint_set_interface(chip, ep, false); if (!--ep->opened) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eadac586bcc8..a50e15be1229 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2185,6 +2185,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ + QUIRK_FLAG_IFACE_SKIP_CLOSE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 2c6575029b1c..e97141ef730a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -170,6 +170,8 @@ extern bool snd_usb_skip_validation; * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) * QUIRK_FLAG_SKIP_IMPLICIT_FB * Don't apply implicit feedback sync mode + * QUIRK_FLAG_IFACE_SKIP_CLOSE + * Don't closed interface during setting sample rate */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -191,5 +193,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) #define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) #define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) +#define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) #endif /* __USBAUDIO_H */ -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Add Positivo C6300 model quirk
by Edson Juliano Drosdeck 10 Nov '22

10 Nov '22

Positivo Master C6300 (1849:a233) require quirk for anabling headset-mic Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701a72ec5629..7875566d6183 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9608,6 +9608,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), + SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- 2.37.2

3 2

[PATCH v6 0/4] Convert soundwire bindings to DT schema
by Srinivasa Rao Mandadapu 10 Nov '22

10 Nov '22

Convert soundwire bindings text file to DT schema and update device tree entries to follow strict dt-bindings. Changes since V5: -- Update indentation of few properties description. -- Update minimum and maximum limit for few properties. -- Fix Typo errors in commit message. -- Update commit message with differences from text file. Changes Since V4: -- Update interrupt names in example. -- Fix blank lines. -- Remove dependecy patch info. -- Split dtsi patches as per SoC. Changes Since V3: -- Remove subnode description and add appropriate pattern properties. -- Add interrput names in example. -- update some properties description. -- Revert minIteams change in previous version. -- Rebase to latest code base. -- Update dtsi node names, -- Remove redundant property in soundwire node. Srinivasa Rao Mandadapu (4): arm64: dts: qcom: Update soundwire secondary node names arm64: dts: qcom: sm8250: Remove redundant soundwire property arm64: dts: qcom: sc7280: Remove redundant soundwire property dt-bindings: soundwire: Convert text bindings to DT Schema .../devicetree/bindings/soundwire/qcom,sdw.txt | 215 ---------------- .../bindings/soundwire/qcom,soundwire.yaml | 270 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sc7280.dtsi | 1 - arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 4 +- .../boot/dts/qcom/sdm850-lenovo-yoga-c630.dts | 4 +- arch/arm64/boot/dts/qcom/sdm850-samsung-w737.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 1 - 9 files changed, 280 insertions(+), 227 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,soundwire.yaml -- 2.7.4

4 7

[PATCH] ASoC: stm32: dfsdm: manage cb buffers cleanup
by Olivier Moysan 09 Nov '22

09 Nov '22

Ensure that resources allocated by iio_channel_get_all_cb() are released on driver unbind. Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> --- sound/soc/stm/stm32_adfsdm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 643fc8a17018..837c1848d9bf 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -304,6 +304,11 @@ static int stm32_adfsdm_dummy_cb(const void *data, void *private) return 0; } +static void stm32_adfsdm_cleanup(void *data) +{ + iio_channel_release_all_cb(data); +} + static struct snd_soc_component_driver stm32_adfsdm_soc_platform = { .open = stm32_adfsdm_pcm_open, .close = stm32_adfsdm_pcm_close, @@ -350,6 +355,12 @@ static int stm32_adfsdm_probe(struct platform_device *pdev) if (IS_ERR(priv->iio_cb)) return PTR_ERR(priv->iio_cb); + ret = devm_add_action_or_reset(&pdev->dev, stm32_adfsdm_cleanup, priv->iio_cb); + if (ret < 0) { + dev_err(&pdev->dev, "Unable to add action\n"); + return ret; + } + component = devm_kzalloc(&pdev->dev, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Add Positivo C6300 model quirk
by Edson Juliano Drosdeck 09 Nov '22

09 Nov '22

Positivo Master C6300 (1849:a233) require quirk for anabling headset-mic Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701a72ec5629..ad2987888966 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9608,6 +9608,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), + SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC233_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- 2.37.2

3 3

[PATCH] CHROMIUM: ASoC: amd: acp: Add tdm support for codecs in machine driver
by Venkata Prasad Potturu 09 Nov '22

09 Nov '22

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Add tdm support for rt5682, rt5682s, rt1019 and nau8825 codec in machine driver. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-mach-common.c | 255 ++++++++++++++++++++++++++-- 1 file changed, 243 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index a78cf29387a7..bce3d8ed48ec 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -27,6 +27,21 @@ #include "../../codecs/nau8825.h" #include "acp-mach.h" +static int tdm_mode = 0; +module_param_named(tdm_mode, tdm_mode, int, 0444); +MODULE_PARM_DESC(tdm_mode, "Set 1 for tdm mode, set 0 for i2s mode"); + +#define TDM_CHANNELS 8 +#define BIT_WIDTH 16 +#define SLOT0 BIT(0) +#define SLOT1 BIT(1) +#define SLOT2 BIT(2) +#define SLOT3 BIT(3) +#define SLOT4 BIT(4) +#define SLOT5 BIT(5) +#define SLOT6 BIT(6) +#define SLOT7 BIT(7) + #define PCO_PLAT_CLK 48000000 #define RT5682_PLL_FREQ (48000 * 512) #define DUAL_CHANNEL 2 @@ -80,13 +95,19 @@ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; int ret; + unsigned int fmt; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); if (drvdata->hs_codec_id != RT5682) return -EINVAL; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP); if (ret < 0) { dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); @@ -151,10 +172,15 @@ static int acp_card_hs_startup(struct snd_pcm_substream *substream) int ret; unsigned int fmt; + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + if (drvdata->soc_mclk) - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; else - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { @@ -191,9 +217,65 @@ static void acp_card_shutdown(struct snd_pcm_substream *substream) clk_disable_unprepare(drvdata->wclk); } +static int acp_card_rt5682x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int ret; + unsigned int fmt; + + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBP_CFP); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 0 and slot 1 for playback and capture. + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, SLOT0 | SLOT1, SLOT0 | SLOT1, + TDM_CHANNELS, BIT_WIDTH); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "set TDM slot err: %d\n", ret); + return ret; + } + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBC_CFC); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 0 and slot 1 for playback and capture. + */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, SLOT0 | SLOT1, SLOT0 | SLOT1, + TDM_CHANNELS, BIT_WIDTH); + if (ret < 0) { + dev_warn(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + } + + return 0; +} + static const struct snd_soc_ops acp_card_rt5682_ops = { .startup = acp_card_hs_startup, .shutdown = acp_card_shutdown, + .hw_params = acp_card_rt5682x_hw_params, }; /* Define RT5682S CODEC component*/ @@ -220,10 +302,15 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) if (drvdata->hs_codec_id != RT5682S) return -EINVAL; + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + if (drvdata->soc_mclk) - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; else - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { @@ -283,6 +370,7 @@ static int acp_card_rt5682s_init(struct snd_soc_pcm_runtime *rtd) static const struct snd_soc_ops acp_card_rt5682s_ops = { .startup = acp_card_hs_startup, + .hw_params = acp_card_rt5682x_hw_params, }; static const unsigned int dmic_channels[] = { @@ -351,19 +439,48 @@ static int acp_card_rt1019_hw_params(struct snd_pcm_substream *substream, struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int srate, i, ret = 0; + unsigned int fmt; srate = params_rate(params); if (drvdata->amp_codec_id != RT1019) return -EINVAL; + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBP_CFP); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 2 and slot 3 for playback. + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, SLOT2 | SLOT3, 0, TDM_CHANNELS, BIT_WIDTH); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "set TDM slot err: %d\n", ret); + return ret; + } + } for_each_rtd_codec_dais(rtd, i, codec_dai) { if (strcmp(codec_dai->name, "rt1019-aif")) continue; - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1019_PLL_S_BCLK, - 64 * srate, 256 * srate); + if (tdm_mode) + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1019_PLL_S_BCLK, + 128 * srate, 256 * srate); + else + ret = snd_soc_dai_set_pll(codec_dai, 0, RT1019_PLL_S_BCLK, + 64 * srate, 256 * srate); + if (ret < 0) return ret; @@ -371,8 +488,36 @@ static int acp_card_rt1019_hw_params(struct snd_pcm_substream *substream, 256 * srate, SND_SOC_CLOCK_IN); if (ret < 0) return ret; - } + if (tdm_mode) { + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A + | SND_SOC_DAIFMT_NB_NF); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + /** + * As codec supports slot 2 for left channel playback. + */ + if (!strcmp(codec_dai->component->name, "i2c-10EC1019:00")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, SLOT2, SLOT2, + TDM_CHANNELS, BIT_WIDTH); + if (ret < 0) + break; + } + + /** + * As codec supports slot 3 for right channel playback. + */ + if (!strcmp(codec_dai->component->name, "i2c-10EC1019:01")) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, SLOT3, SLOT3, + TDM_CHANNELS, BIT_WIDTH); + if (ret < 0) + break; + } + } + } return 0; } @@ -426,9 +571,43 @@ static int acp_card_maxim_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(max98360a_map)); } +static int acp_card_maxim_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + unsigned int fmt; + int ret; + + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBP_CFP); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 2 and slot 3 for playback. + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, SLOT2 | SLOT3, 0, TDM_CHANNELS, BIT_WIDTH); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "set TDM slot err: %d\n", ret); + return ret; + } + } + return 0; +} + static const struct snd_soc_ops acp_card_maxim_ops = { .startup = acp_card_amp_startup, .shutdown = acp_card_shutdown, + .hw_params = acp_card_maxim_hw_params, }; /* Declare nau8825 codec components */ @@ -454,10 +633,15 @@ static int acp_card_nau8825_init(struct snd_soc_pcm_runtime *rtd) if (drvdata->hs_codec_id != NAU8825) return -EINVAL; + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + if (drvdata->soc_mclk) - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; else - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; + fmt = fmt | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP; ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { @@ -493,8 +677,34 @@ static int acp_nau8825_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); int ret; + unsigned int fmt; + if (tdm_mode) + fmt = SND_SOC_DAIFMT_DSP_A; + else + fmt = SND_SOC_DAIFMT_I2S; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBP_CFP); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 4 and slot 5 for playback + * and slot 6 and slot 7 for capture. + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, SLOT4 | SLOT5, SLOT6 | SLOT7, + TDM_CHANNELS, BIT_WIDTH); + if (ret && ret != -ENOTSUPP) { + dev_err(rtd->dev, "set TDM slot err: %d\n", ret); + return ret; + } + } ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, (48000 * 256), SND_SOC_CLOCK_IN); if (ret < 0) @@ -507,6 +717,25 @@ static int acp_nau8825_hw_params(struct snd_pcm_substream *substream, return ret; } + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBC_CFC); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set dai fmt: %d\n", ret); + return ret; + } + + if (tdm_mode) { + /** + * As codec supports slot 4 and slot 5 for playback and slot 6 for capture. + */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, SLOT6, SLOT4 | SLOT5, + TDM_CHANNELS, BIT_WIDTH); + if (ret < 0) { + dev_warn(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + } + return ret; } @@ -567,6 +796,8 @@ SND_SOC_DAILINK_DEF(sof_sp, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-sp"))); SND_SOC_DAILINK_DEF(sof_hs, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-hs"))); +SND_SOC_DAILINK_DEF(sof_hs_virtual, + DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-hs-virtual"))); SND_SOC_DAILINK_DEF(sof_dmic, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-dmic"))); SND_SOC_DAILINK_DEF(pdm_dmic, @@ -733,8 +964,8 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->amp_cpu_id == I2S_HS) { links[i].name = "acp-amp-codec"; links[i].id = AMP_BE_ID; - links[i].cpus = sof_hs; - links[i].num_cpus = ARRAY_SIZE(sof_hs); + links[i].cpus = sof_hs_virtual; + links[i].num_cpus = ARRAY_SIZE(sof_hs_virtual); links[i].platforms = sof_component; links[i].num_platforms = ARRAY_SIZE(sof_component); links[i].dpcm_playback = 1; -- 2.25.1

4 11

[PATCH] pcm: rate - check rate type for using snd_pcm_rate_slave_frames
by Shengjiu Wang 09 Nov '22

09 Nov '22

With plughw device and mmap case, the plug pcm fast_ops pointer is same as slave pcm fast_ops, but ops pointer is different, which cause the "bus error" in snd_pcm_rate_slave_frames. The test command is arecord -Dplughw:x -r12000 -c2 -fS16_LE -M temp.wav This patch is to add pcm type check as commit: d21e0e01 pcm: plugin - fix avail_min calculation on rate plugin Fixes: d9dbb57b ("pcm: rate - rewrite the may_wait_for_avail_min callback for the rate plugin") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- src/pcm/pcm_rate.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index e8815e8b..dc502202 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1304,8 +1304,11 @@ static snd_pcm_uframes_t snd_pcm_rate_slave_frames(snd_pcm_t *pcm, snd_pcm_ufram static int snd_pcm_rate_may_wait_for_avail_min(snd_pcm_t *pcm, snd_pcm_uframes_t avail) { - return snd_pcm_plugin_may_wait_for_avail_min_conv(pcm, avail, - snd_pcm_rate_slave_frames); + if (snd_pcm_type(pcm) == SND_PCM_TYPE_RATE) + return snd_pcm_plugin_may_wait_for_avail_min_conv(pcm, avail, + snd_pcm_rate_slave_frames); + else + return snd_pcm_plugin_may_wait_for_avail_min_conv(pcm, avail, NULL); } static const snd_pcm_fast_ops_t snd_pcm_rate_fast_ops = { -- 2.34.1

3 2

[PATCH v3 0/3] arm64/slimbus/dt-bindings: convert to DT Schema, minor cleanups
by Krzysztof Kozlowski 09 Nov '22

09 Nov '22

Hi, Changes since v2 ================ 1. None, rebased and dropped applied patches. Changes since v1 ================ 1. Fix commit title typo (Steev). 2. Add Rb/Tb tags. Dependencies ============ No dependencies. Binding patches are independent from DTS. Best regards, Krzysztof Krzysztof Kozlowski (3): dt-bindings: slimbus: convert bus description to DT schema dt-bindings: slimbus: qcom,slim: convert to DT schema dt-bindings: slimbus: qcom,slim-ngd: convert to DT schema .../devicetree/bindings/slimbus/bus.txt | 60 --------- .../bindings/slimbus/qcom,slim-ngd.yaml | 120 ++++++++++++++++++ .../bindings/slimbus/qcom,slim.yaml | 86 +++++++++++++ .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 84 ------------ .../bindings/slimbus/slim-qcom-ctrl.txt | 39 ------ .../devicetree/bindings/slimbus/slimbus.yaml | 95 ++++++++++++++ 6 files changed, 301 insertions(+), 183 deletions(-) delete mode 100644 Documentation/devicetree/bindings/slimbus/bus.txt create mode 100644 Documentation/devicetree/bindings/slimbus/qcom,slim-ngd.yaml create mode 100644 Documentation/devicetree/bindings/slimbus/qcom,slim.yaml delete mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt delete mode 100644 Documentation/devicetree/bindings/slimbus/slim-qcom-ctrl.txt create mode 100644 Documentation/devicetree/bindings/slimbus/slimbus.yaml -- 2.34.1

2 5

[PATCH 0/4] Set BQ parameters for some Dell models
by shumingf＠realtek.com 09 Nov '22

09 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> There are some Dell SKUs that need to set the parameters of the crossover filter (biquad). Each amplifier connects to one tweeter speaker and one woofer speaker. We should control HPF/LPF to output the proper frequency for the different speakers. If the codec driver got the BQ parameters from the device property, it will apply these parameters to the hardware. Shuming Fan (4): ASoC: rt1308-sdw: get BQ params property and apply them ASoC: rt1316-sdw: get BQ params property and apply them ASoC: Intel: sof_sdw_rt1308: add BQ params for the Dell models ASoC: Intel: sof_sdw_rt1316: add BQ params for the Dell models sound/soc/codecs/rt1308-sdw.c | 39 +++ sound/soc/codecs/rt1308-sdw.h | 2 + sound/soc/codecs/rt1316-sdw.c | 39 +++ sound/soc/codecs/rt1316-sdw.h | 2 + sound/soc/intel/boards/sof_sdw.c | 2 + .../intel/boards/sof_sdw_amp_coeff_tables.h | 300 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 3 + sound/soc/intel/boards/sof_sdw_rt1308.c | 120 +++++++ sound/soc/intel/boards/sof_sdw_rt1316.c | 119 +++++++ 9 files changed, 626 insertions(+) create mode 100644 sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h -- 2.38.1

1 0

[PATCH] ALSA: usb-audio: Add DSD support for Accuphase DAC-60
by Jussi Laako 09 Nov '22

09 Nov '22

Accuphase DAC-60 option card supports native DSD up to DSD256, but doesn't have support for auto-detection. Explicitly enable DSD support for the correct altsetting. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2dd19a5c3026..aedc3987a4cf 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1875,6 +1875,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ + case USB_ID(0x21ed, 0xd75a): /* Accuphase DAC-60 option card */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) -- 2.34.1

2 1

[PATCH 0/2] soundwire: cadence: remove dma_data
by Bard Liao 09 Nov '22

09 Nov '22

The use of dma_data is problematic for two reasons for the Cadence IP. a) the dai runtime data has nothing to do with DMAs in existing solutions b) we will use the dma_data for DMA-management in the future. We cannot share two separate pieces of information with the same dma_data pointer. Pierre-Louis Bossart (2): soundwire: cadence: rename sdw_cdns_dai_dma_data as sdw_cdns_dai_runtime soundwire: cadence: use dai_runtime_array instead of dma_data drivers/soundwire/cadence_master.c | 50 +++++++------ drivers/soundwire/cadence_master.h | 9 ++- drivers/soundwire/intel.c | 111 ++++++++++++++--------------- 3 files changed, 86 insertions(+), 84 deletions(-) -- 2.25.1

2 3

[PATCH 0/3] ASoC: core: Suspend/resume error propagation
by Cezary Rojewski 08 Nov '22

08 Nov '22

Currently all erros encountered during suspend/resume are squelched what limits the error-handling possibilities. Patches found here combined together allow for those errors to be propagated up. This aligns ASoC component's behavior with how PM-framework handles suspend/resume for a device - there's always an option for a client (driver) to return an error code in case of failure. Amadeusz Sławiński (3): ASoC: component: Propagate result of suspend and resume callbacks ASoC: core: Inline resume work back to resume function ASoC: core: Propagate component suspend/resume errors include/sound/soc-component.h | 4 +- include/sound/soc.h | 3 -- sound/soc/soc-component.c | 22 +++++++--- sound/soc/soc-core.c | 75 ++++++++++++++++------------------- 4 files changed, 52 insertions(+), 52 deletions(-) -- 2.25.1

4 11

Re: [PATCH linux-next] ASoC: SOF: remove duplicated included sof-audio.h
by Pierre-Louis Bossart 08 Nov '22

08 Nov '22

You need to CC: maintainers and the alsa-devel mailing list in a v2 On 11/8/22 05:31, yang.yang29(a)zte.com.cn wrote: > From: Xu Panda <xu.panda(a)zte.com.cn> > > The sof-audio.h is included more than once. > > Reported-by: Zeal Robot <zealci(a)zte.com.cn> > Signed-off-by: Xu Panda <xu.panda(a)zte.com.cn> > Signed-off-by: Yang Yang <yang.yang29(a)zte.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > --- > sound/soc/sof/amd/acp-common.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c > index 27b95187356e..348e70dfe2a5 100644 > --- a/sound/soc/sof/amd/acp-common.c > +++ b/sound/soc/sof/amd/acp-common.c > @@ -13,7 +13,6 @@ > #include "../sof-priv.h" > #include "../sof-audio.h" > #include "../ops.h" > -#include "../sof-audio.h" > #include "acp.h" > #include "acp-dsp-offset.h" >

1 0

[PATCH] ALSA: usb-audio: Add quirk entry for M-Audio Micro
by Takashi Iwai 08 Nov '22

08 Nov '22

M-Audio Micro (0762:201a) defines the descriptor as vendor-specific, while the content seems class-compliant. Just overriding the probe makes the device working. Reported-by: Ash Logan <ash(a)heyquark.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/7ecd4417-d860-4773-c1c1-b07433342390@heyquark.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 06dfdd45cff8..874fcf245747 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2049,6 +2049,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* M-Audio Micro */ + USB_DEVICE_VENDOR_SPEC(0x0763, 0x201a), +}, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2030), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { -- 2.35.3

1 0

[PATCH v5 0/4] Convert soundwire bindings to DT schema
by Srinivasa Rao Mandadapu 08 Nov '22

08 Nov '22

Convert soundwire bindings text file to DT schema and update device tree entries to follow strict dt-bindings. Changes Since V4: -- Update interrupt names in example. -- Fix blank lines. -- Remove dependecy patch info. -- Split dtsi patches as per SoC. Changes Since V3: -- Remove subnode description and add appropriate pattern properties. -- Add interrput names in example. -- update some properties description. -- Revert minIteams change in previous version. -- Rebase to latest code base. -- Update dtsi node names, -- Remove redundant property in soundwire node. Srinivasa Rao Mandadapu (4): arm64: dts: qcom: Update soundwire slave node names arm64: dts: qcom: sm8250: Remove redundant soundwire property arm64: dts: qcom: sc7280: Remove redundant soundwire property dt-bindings: soundwire: Convert text bindings to DT Schema .../devicetree/bindings/soundwire/qcom,sdw.txt | 215 ------------------ .../bindings/soundwire/qcom,soundwire.yaml | 244 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sc7280.dtsi | 1 - arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 4 +- .../boot/dts/qcom/sdm850-lenovo-yoga-c630.dts | 4 +- arch/arm64/boot/dts/qcom/sdm850-samsung-w737.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 1 - 9 files changed, 254 insertions(+), 227 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,soundwire.yaml -- 2.7.4

2 10

[PATCH] [RFC] selftests: alsa - add PCM test
by Jaroslav Kysela 08 Nov '22

08 Nov '22

This initial code does a simple sample transfer tests. By default, all PCM devices are detected and tested with short and long buffering parameters for 4 seconds. If the sample transfer timing is not in a +-100ms boundary, the test fails. Only the interleaved buffering scheme is supported in this version. The configuration may be modified with the configuration files. A specific hardware configuration is detected and activated using the sysfs regex matching. This allows to use the DMI string (/sys/class/dmi/id/* tree) or any other system parameters exposed in sysfs for the matching for the CI automation. The configuration file may also specify the PCM device list to detect the missing PCM devices. Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)intel.com> Cc: Jesse Barnes <jsbarnes(a)google.com> Cc: Jimmy Cheng-Yi Chiang <cychiang(a)google.com> Cc: Curtis Malainey <cujomalainey(a)google.com> Cc: Brian Norris <briannorris(a)chromium.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- tools/testing/selftests/alsa/Makefile | 4 +- tools/testing/selftests/alsa/conf.c | 379 +++++++++++++ .../alsa/conf.d/Lenovo_ThinkPad_P1_Gen2.conf | 76 +++ tools/testing/selftests/alsa/pcm-test.c | 523 ++++++++++++++++++ 4 files changed, 981 insertions(+), 1 deletion(-) create mode 100644 tools/testing/selftests/alsa/conf.c create mode 100644 tools/testing/selftests/alsa/conf.d/Lenovo_ThinkPad_P1_Gen2.conf create mode 100644 tools/testing/selftests/alsa/pcm-test.c diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index fd8ddce2b1a6..8ac22d2eb2ba 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -7,6 +7,8 @@ ifeq ($(LDLIBS),) LDLIBS += -lasound endif -TEST_GEN_PROGS := mixer-test +TEST_GEN_PROGS := mixer-test pcm-test + +pcm-test: pcm-test.c conf.c include ../lib.mk diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c new file mode 100644 index 000000000000..dd6aa90deef9 --- /dev/null +++ b/tools/testing/selftests/alsa/conf.c @@ -0,0 +1,379 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// kselftest configuration helpers for the hw specific configuration +// +// Original author: Jaroslav Kysela <perex(a)perex.cz> +// Copyright (c) 2022 Red Hat Inc. + +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <errno.h> +#include <assert.h> +#include <dirent.h> +#include <regex.h> +#include <sys/stat.h> + +#include "../kselftest.h" +#include "alsa-local.h" + +#define SYSFS_ROOT "/sys" + +struct card_data { + int card; + snd_config_t *config; + const char *filename; + struct card_data *next; +}; + +static struct card_data *conf_cards; + +static struct card_data *conf_data_by_card(int card, bool msg) +{ + struct card_data *conf; + + for (conf = conf_cards; conf; conf = conf->next) { + if (conf->card == card) { + if (msg) + ksft_print_msg("using hw card config %s for card %d\n", + conf->filename, card); + return conf; + } + } + return NULL; +} + +static int dump_config_tree(snd_config_t *top) +{ + snd_output_t *out; + int err; + + err = snd_output_stdio_attach(&out, stdout, 0); + if (err < 0) + ksft_exit_fail_msg("stdout attach\n"); + if (snd_config_save(top, out)) + ksft_exit_fail_msg("config save\n"); + snd_output_close(out); +} + +static snd_config_t *load(const char *filename) +{ + snd_config_t *dst; + snd_input_t *input; + int err; + + err = snd_input_stdio_open(&input, filename, "r"); + if (err < 0) + ksft_exit_fail_msg("Unable to parse filename %s\n", filename); + err = snd_config_top(&dst); + if (err < 0) + ksft_exit_fail_msg("Out of memory\n"); + err = snd_config_load(dst, input); + snd_input_close(input); + if (err < 0) + ksft_exit_fail_msg("Unable to parse filename %s\n", filename); + return dst; +} + +static char *sysfs_get(const char *sysfs_root, const char *id) +{ + char path[PATH_MAX], link[PATH_MAX + 1]; + struct stat sb; + ssize_t len; + char *e; + int fd; + + if (id[0] == '/') + id++; + snprintf(path, sizeof(path), "%s/%s", sysfs_root, id); + if (lstat(path, &sb) != 0) + return NULL; + if (S_ISLNK(sb.st_mode)) { + len = readlink(path, link, sizeof(link) - 1); + if (len <= 0) { + ksft_exit_fail_msg("sysfs: cannot read link '%s': %s\n", + path, strerror(errno)); + return NULL; + } + link[len] = '\0'; + e = strrchr(link, '/'); + if (e) + return strdup(e + 1); + return NULL; + } + if (S_ISDIR(sb.st_mode)) + return NULL; + if ((sb.st_mode & S_IRUSR) == 0) + return NULL; + + fd = open(path, O_RDONLY); + if (fd < 0) { + if (errno == ENOENT) + return NULL; + ksft_exit_fail_msg("sysfs: open failed for '%s': %s\n", + path, strerror(errno)); + } + len = read(fd, path, sizeof(path)-1); + close(fd); + if (len < 0) + ksft_exit_fail_msg("sysfs: unable to read value '%s': %s\n", + path, errno); + while (len > 0 && path[len-1] == '\n') + len--; + path[len] = '\0'; + e = strdup(path); + if (e == NULL) + ksft_exit_fail_msg("Out of memory\n"); + return e; +} + +static bool sysfs_match(const char *sysfs_root, snd_config_t *config) +{ + snd_config_t *node, *path_config, *regex_config; + snd_config_iterator_t i, next; + const char *path_string, *regex_string, *v; + regex_t re; + regmatch_t match[1]; + int iter = 0, ret; + + snd_config_for_each(i, next, config) { + node = snd_config_iterator_entry(i); + if (snd_config_search(node, "path", &path_config)) + ksft_exit_fail_msg("Missing path field in the sysfs block\n"); + if (snd_config_search(node, "regex", &regex_config)) + ksft_exit_fail_msg("Missing regex field in the sysfs block\n"); + if (snd_config_get_string(path_config, &path_string)) + ksft_exit_fail_msg("Path field in the sysfs block is not a string\n"); + if (snd_config_get_string(regex_config, &regex_string)) + ksft_exit_fail_msg("Regex field in the sysfs block is not a string\n"); + iter++; + v = sysfs_get(sysfs_root, path_string); + if (!v) + return false; + if (regcomp(&re, regex_string, REG_EXTENDED)) + ksft_exit_fail_msg("Wrong regex '%s'\n", regex_string); + ret = regexec(&re, v, 1, match, 0); + regfree(&re); + if (ret) + return false; + } + return iter > 0; +} + +static bool test_filename1(int card, const char *filename, const char *sysfs_card_root) +{ + struct card_data *data, *data2; + snd_config_t *config, *sysfs_config, *card_config, *sysfs_card_config, *node; + snd_config_iterator_t i, next; + + config = load(filename); + if (snd_config_search(config, "sysfs", &sysfs_config) || + snd_config_get_type(sysfs_config) != SND_CONFIG_TYPE_COMPOUND) + ksft_exit_fail_msg("Missing global sysfs block in filename %s\n", filename); + if (snd_config_search(config, "card", &card_config) || + snd_config_get_type(card_config) != SND_CONFIG_TYPE_COMPOUND) + ksft_exit_fail_msg("Missing global card block in filename %s\n", filename); + if (!sysfs_match(SYSFS_ROOT, sysfs_config)) + return false; + snd_config_for_each(i, next, card_config) { + node = snd_config_iterator_entry(i); + if (snd_config_search(node, "sysfs", &sysfs_card_config) || + snd_config_get_type(sysfs_card_config) != SND_CONFIG_TYPE_COMPOUND) + ksft_exit_fail_msg("Missing card sysfs block in filename %s\n", filename); + if (!sysfs_match(sysfs_card_root, sysfs_card_config)) + continue; + data = malloc(sizeof(*data)); + if (!data) + ksft_exit_fail_msg("Out of memory\n"); + data2 = conf_data_by_card(card, false); + if (data2) + ksft_exit_fail_msg("Duplicate card '%s' <-> '%s'\n", filename, data2->filename); + data->card = card; + data->filename = filename; + data->config = node; + data->next = conf_cards; + conf_cards = data; + return true; + } + return false; +} + +static bool test_filename(const char *filename) +{ + char fn[128]; + int card; + + for (card = 0; card < 32; card++) { + snprintf(fn, sizeof(fn), "%s/class/sound/card%d", SYSFS_ROOT, card); + if (access(fn, R_OK) == 0 && test_filename1(card, filename, fn)) + return true; + } + return false; +} + +static int filename_filter(const struct dirent *dirent) +{ + size_t flen; + + if (dirent == NULL) + return 0; + if (dirent->d_type == DT_DIR) + return 0; + flen = strlen(dirent->d_name); + if (flen <= 5) + return 0; + if (strncmp(&dirent->d_name[flen-5], ".conf", 5) == 0) + return 1; + return 0; +} + +void conf_load(void) +{ + const char *fn = "conf.d"; + struct dirent **namelist; + int n, j; + + n = scandir(fn, &namelist, filename_filter, alphasort); + if (n < 0) + ksft_exit_fail_msg("scandir: %s\n", strerror(errno)); + for (j = 0; j < n; j++) { + size_t sl = strlen(fn) + strlen(namelist[j]->d_name) + 2; + char *filename = malloc(sl); + if (filename == NULL) + ksft_exit_fail_msg("Out of memory\n"); + sprintf(filename, "%s/%s", fn, namelist[j]->d_name); + if (test_filename(filename)) + filename = NULL; + free(filename); + free(namelist[j]); + } + free(namelist); +} + +void conf_free(void) +{ + struct card_data *conf; + + while (conf_cards) { + conf = conf_cards; + conf_cards = conf->next; + snd_config_delete(conf->config); + } +} + +snd_config_t *conf_by_card(int card) +{ + struct card_data *conf; + + conf = conf_data_by_card(card, true); + if (conf) + return conf->config; + return NULL; +} + +static int conf_get_by_keys(snd_config_t *root, const char *key1, + const char *key2, snd_config_t **result) +{ + int ret; + + if (key1) { + ret = snd_config_search(root, key1, &root); + if (ret != -ENOENT && ret < 0) + return ret; + } + if (key2) + ret = snd_config_search(root, key2, &root); + if (ret >= 0) + *result = root; + return ret; +} + +snd_config_t *conf_get_subtree(snd_config_t *root, const char *key1, const char *key2) +{ + int ret; + + if (!root) + return NULL; + ret = conf_get_by_keys(root, key1, key2, &root); + if (ret == -ENOENT) + return NULL; + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); + return root; +} + +int conf_get_count(snd_config_t *root, const char *key1, const char *key2) +{ + snd_config_t *cfg; + snd_config_iterator_t i, next; + int count, ret; + + if (!root) + return -1; + ret = conf_get_by_keys(root, key1, key2, &cfg); + if (ret == -ENOENT) + return -1; + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); + if (snd_config_get_type(cfg) != SND_CONFIG_TYPE_COMPOUND) + ksft_exit_fail_msg("key '%s'.'%s' is not a compound\n", key1, key2); + count = 0; + snd_config_for_each(i, next, cfg) + count++; + return count; +} + +const char *conf_get_string(snd_config_t *root, const char *key1, const char *key2, const char *def) +{ + snd_config_t *cfg; + const char *s; + int ret; + + if (!root) + return def; + ret = conf_get_by_keys(root, key1, key2, &cfg); + if (ret == -ENOENT) + return def; + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); + if (snd_config_get_string(cfg, &s)) + ksft_exit_fail_msg("key '%s'.'%s' is not a string\n", key1, key2); + return s; +} + +long conf_get_long(snd_config_t *root, const char *key1, const char *key2, long def) +{ + snd_config_t *cfg; + long l; + int ret; + + if (!root) + return def; + ret = conf_get_by_keys(root, key1, key2, &cfg); + if (ret == -ENOENT) + return def; + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); + if (snd_config_get_integer(cfg, &l)) + ksft_exit_fail_msg("key '%s'.'%s' is not an integer\n", key1, key2); + return l; +} + +int conf_get_bool(snd_config_t *root, const char *key1, const char *key2, int def) +{ + snd_config_t *cfg; + long l; + int ret; + + if (!root) + return def; + ret = conf_get_by_keys(root, key1, key2, &cfg); + if (ret == -ENOENT) + return def; + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); + ret = snd_config_get_bool(cfg); + if (ret < 0) + ksft_exit_fail_msg("key '%s'.'%s' is not an bool\n", key1, key2); + return !!ret; +} diff --git a/tools/testing/selftests/alsa/conf.d/Lenovo_ThinkPad_P1_Gen2.conf b/tools/testing/selftests/alsa/conf.d/Lenovo_ThinkPad_P1_Gen2.conf new file mode 100644 index 000000000000..0a83f35d43eb --- /dev/null +++ b/tools/testing/selftests/alsa/conf.d/Lenovo_ThinkPad_P1_Gen2.conf @@ -0,0 +1,76 @@ +# +# Example configuration for Lenovo ThinkPad P1 Gen2 +# + +# +# Use regex match for the string read from the given sysfs path +# +# The sysfs root directory (/sys) is hardwired in the test code +# (may be changed on demand). +# +# All strings must match. +# +sysfs [ + { + path "class/dmi/id/product_sku" + regex "LENOVO_MT_20QU_BU_Think_FM_ThinkPad P1 Gen 2" + } +] + +card.hda { + # + # Use regex match for the /sys/class/sound/card*/ tree (relative) + # + sysfs [ + { + path "device/subsystem_device" + regex "0x229e" + } + { + path "device/subsystem_vendor" + regex "0x17aa" + } + ] + + # + # PCM configuration + # + # pcm.0.0 - device 0 subdevice 0 + # + pcm.0.0 { + PLAYBACK { + test.time1 { + access RW_INTERLEAVED # can be omitted - default + format S16_LE # can be omitted - default + rate 48000 # can be omitted - default + channels 2 # can be omitted - default + period_size 512 + buffer_size 4096 + } + test.time2 { + access RW_INTERLEAVED + format S16_LE + rate 48000 + channels 2 + period_size 24000 + buffer_size 192000 + } + } + CAPTURE { + # use default tests, check for the presence + } + } + # + # uncomment to force the missing device checks + # + #pcm.0.2 { + # PLAYBACK { + # # check for the presence + # } + #} + #pcm.0.3 { + # CAPTURE { + # # check for the presence + # } + #} +} diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c new file mode 100644 index 000000000000..2814d8f74f82 --- /dev/null +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -0,0 +1,523 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// kselftest for the ALSA PCM API +// +// Original author: Jaroslav Kysela <perex(a)perex.cz> +// Copyright (c) 2022 Red Hat Inc. + +// This test will iterate over all cards detected in the system, exercising +// every PCM device it can find. This may conflict with other system +// software if there is audio activity so is best run on a system with a +// minimal active userspace. + +#include <stdio.h> +#include <stdlib.h> +#include <stdbool.h> +#include <errno.h> +#include <assert.h> + +#include "../kselftest.h" +#include "alsa-local.h" + +typedef struct timespec timestamp_t; + +struct pcm_data { + snd_pcm_t *handle; + int card; + int device; + int subdevice; + snd_pcm_stream_t stream; + snd_config_t *pcm_config; + struct pcm_data *next; +}; + +static const char *alsa_config = +"ctl.hw {\n" +" @args [ CARD ]\n" +" @args.CARD.type string\n" +" type hw\n" +" card $CARD\n" +"}\n" +"pcm.hw {\n" +" @args [ CARD DEV SUBDEV ]\n" +" @args.CARD.type string\n" +" @args.DEV.type integer\n" +" @args.SUBDEV.type integer\n" +" type hw\n" +" card $CARD\n" +" device $DEV\n" +" subdevice $SUBDEV\n" +"}\n" + +; + +int num_pcms = 0; +struct pcm_data *pcm_list = NULL; + +int num_missing = 0; +struct pcm_data *pcm_missing = NULL; + +void timestamp_now(timestamp_t *tstamp) +{ + if (clock_gettime(CLOCK_MONOTONIC_RAW, tstamp)) + ksft_exit_fail_msg("clock_get_time\n"); +} + +long long timestamp_diff_ms(timestamp_t *tstamp) +{ + timestamp_t now, diff; + timestamp_now(&now); + if (tstamp->tv_nsec > now.tv_nsec) { + diff.tv_sec = now.tv_sec - tstamp->tv_sec - 1; + diff.tv_nsec = (now.tv_nsec + 1000000000L) - tstamp->tv_nsec; + } else { + diff.tv_sec = now.tv_sec - tstamp->tv_sec; + diff.tv_nsec = now.tv_nsec - tstamp->tv_nsec; + } + return (diff.tv_sec * 1000) + ((diff.tv_nsec + 500000L) / 1000000L); +} + +#ifdef SND_LIB_VER +#if SND_LIB_VERSION >= SND_LIB_VER(1, 2, 6) +#define LIB_HAS_LOAD_STRING +#endif +#endif + +#ifndef LIB_HAS_LOAD_STRING +static int snd_config_load_string(snd_config_t **config, const char *s, + size_t size) +{ + snd_input_t *input; + snd_config_t *dst; + int err; + + assert(config && s); + if (size == 0) + size = strlen(s); + err = snd_input_buffer_open(&input, s, size); + if (err < 0) + return err; + err = snd_config_top(&dst); + if (err < 0) { + snd_input_close(input); + return err; + } + err = snd_config_load(dst, input); + snd_input_close(input); + if (err < 0) { + snd_config_delete(dst); + return err; + } + *config = dst; + return 0; +} +#endif + +static snd_config_t *get_alsalib_config(void) +{ + snd_config_t *config; + int err; + + err = snd_config_load_string(&config, alsa_config, strlen(alsa_config)); + if (err < 0) { + ksft_print_msg("Unable to parse custom alsa-lib configuration: %s\n", + snd_strerror(err)); + ksft_exit_fail(); + } + return config; +} + +static long device_from_id(snd_config_t *node) +{ + const char *id; + char *end; + long v; + + if (snd_config_get_id(node, &id)) + ksft_exit_fail_msg("snd_config_get_id\n"); + errno = 0; + v = strtol(id, &end, 10); + if (errno || *end) + return -1; + return v; +} + +static void missing_device(int card, int device, int subdevice, snd_pcm_stream_t stream) +{ + struct pcm_data *pcm_data; + + for (pcm_data = pcm_list; pcm_data != NULL; pcm_data = pcm_data->next) { + if (pcm_data->card != card) + continue; + if (pcm_data->device != device) + continue; + if (pcm_data->subdevice != subdevice) + continue; + if (pcm_data->stream != stream) + continue; + return; + } + pcm_data = calloc(1, sizeof(*pcm_data)); + if (!pcm_data) + ksft_exit_fail_msg("Out of memory\n"); + pcm_data->card = card; + pcm_data->device = device; + pcm_data->subdevice = subdevice; + pcm_data->stream = stream; + pcm_data->next = pcm_missing; + pcm_missing = pcm_data; + num_missing++; +} + +static void missing_devices(int card, snd_config_t *card_config) +{ + snd_config_t *pcm_config, *node1, *node2; + snd_config_iterator_t i1, i2, next1, next2; + int device, subdevice; + + pcm_config = conf_get_subtree(card_config, "pcm", NULL); + if (!pcm_config) + return; + snd_config_for_each(i1, next1, pcm_config) { + node1 = snd_config_iterator_entry(i1); + device = device_from_id(node1); + if (device < 0) + continue; + if (snd_config_get_type(node1) != SND_CONFIG_TYPE_COMPOUND) + continue; + snd_config_for_each(i2, next2, node1) { + node2 = snd_config_iterator_entry(i2); + subdevice = device_from_id(node2); + if (subdevice < 0) + continue; + if (conf_get_subtree(node2, "PLAYBACK", NULL)) + missing_device(card, device, subdevice, SND_PCM_STREAM_PLAYBACK); + if (conf_get_subtree(node2, "CAPTURE", NULL)) + missing_device(card, device, subdevice, SND_PCM_STREAM_CAPTURE); + } + } +} + +static void find_pcms(void) +{ + char name[32], key[64]; + int card, dev, subdev, count, direction, err; + snd_pcm_stream_t stream; + struct pcm_data *pcm_data; + snd_ctl_t *handle; + snd_pcm_info_t *pcm_info; + snd_config_t *config, *card_config, *pcm_config; + + snd_pcm_info_alloca(&pcm_info); + + card = -1; + if (snd_card_next(&card) < 0 || card < 0) + return; + + config = get_alsalib_config(); + + while (card >= 0) { + sprintf(name, "hw:%d", card); + + err = snd_ctl_open_lconf(&handle, name, 0, config); + if (err < 0) { + ksft_print_msg("Failed to get hctl for card %d: %s\n", + card, snd_strerror(err)); + goto next_card; + } + + card_config = conf_by_card(card); + + dev = -1; + while (1) { + if (snd_ctl_pcm_next_device(handle, &dev) < 0) + ksft_exit_fail_msg("snd_ctl_pcm_next_device\n"); + if (dev < 0) + break; + + for (direction = 0; direction < 2; direction++) { + stream = direction ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK; + sprintf(key, "pcm.%d.%s", dev, snd_pcm_stream_name(stream)); + pcm_config = conf_get_subtree(card_config, key, NULL); + if (conf_get_bool(card_config, key, "skip", false)) { + ksft_print_msg("skipping pcm %d.%d.%s\n", card, dev, snd_pcm_stream_name(stream)); + continue; + } + snd_pcm_info_set_device(pcm_info, dev); + snd_pcm_info_set_subdevice(pcm_info, 0); + snd_pcm_info_set_stream(pcm_info, stream); + err = snd_ctl_pcm_info(handle, pcm_info); + if (err == -ENOENT) + continue; + if (err < 0) + ksft_exit_fail_msg("snd_ctl_pcm_info: %d:%d:%d\n", + dev, 0, stream); + count = snd_pcm_info_get_subdevices_count(pcm_info); + for (subdev = 0; subdev < count; subdev++) { + sprintf(key, "pcm.%d.%d.%s", dev, subdev, snd_pcm_stream_name(stream)); + if (conf_get_bool(card_config, key, "skip", false)) { + ksft_print_msg("skipping pcm %d.%d.%d.%s\n", card, dev, + subdev, snd_pcm_stream_name(stream)); + continue; + } + pcm_data = calloc(1, sizeof(*pcm_data)); + if (!pcm_data) + ksft_exit_fail_msg("Out of memory\n"); + pcm_data->card = card; + pcm_data->device = dev; + pcm_data->subdevice = subdev; + pcm_data->stream = stream; + pcm_data->pcm_config = conf_get_subtree(card_config, key, NULL); + pcm_data->next = pcm_list; + pcm_list = pcm_data; + num_pcms++; + } + } + } + + /* check for missing devices */ + missing_devices(card, card_config); + + next_card: + snd_ctl_close(handle); + if (snd_card_next(&card) < 0) { + ksft_print_msg("snd_card_next"); + break; + } + } + + snd_config_delete(config); +} + +static void test_pcm_time1(struct pcm_data *data, + const char *cfg_prefix, const char *sformat, + long srate, long schannels, + long speriod_size, long sbuffer_size) +{ + char name[64], key[128], msg[256]; + const char *cs; + int i, err; + snd_pcm_t *handle = NULL; + snd_pcm_access_t access = SND_PCM_ACCESS_RW_INTERLEAVED; + snd_pcm_format_t format; + unsigned char *samples = NULL; + snd_pcm_sframes_t frames; + long long ms; + long rate, channels, period_size, buffer_size; + unsigned int rrate; + snd_pcm_uframes_t rperiod_size, rbuffer_size, start_threshold; + timestamp_t tstamp; + bool pass = false, automatic = true; + snd_pcm_hw_params_t *hw_params; + snd_pcm_sw_params_t *sw_params; + + snd_pcm_hw_params_alloca(&hw_params); + snd_pcm_sw_params_alloca(&sw_params); + + cs = conf_get_string(data->pcm_config, cfg_prefix, "format", sformat); + format = snd_pcm_format_value(cs); + if (format == SND_PCM_FORMAT_UNKNOWN) + ksft_exit_fail_msg("Wrong format '%s'\n", cs); + rate = conf_get_long(data->pcm_config, cfg_prefix, "rate", srate); + channels = conf_get_long(data->pcm_config, cfg_prefix, "channels", schannels); + period_size = conf_get_long(data->pcm_config, cfg_prefix, "period_size", speriod_size); + buffer_size = conf_get_long(data->pcm_config, cfg_prefix, "buffer_size", sbuffer_size); + + automatic = strcmp(sformat, snd_pcm_format_name(format)) == 0 && + srate == rate && + schannels == channels && + speriod_size == period_size && + sbuffer_size == buffer_size; + + samples = malloc((rate * channels * snd_pcm_format_physical_width(format)) / 8); + if (!samples) + ksft_exit_fail_msg("Out of memory\n"); + snd_pcm_format_set_silence(format, samples, rate * channels); + + sprintf(name, "hw:%d,%d,%d", data->card, data->device, data->subdevice); + err = snd_pcm_open(&handle, name, data->stream, 0); + if (err < 0) { + snprintf(msg, sizeof(msg), "Failed to get pcm handle: %s", snd_strerror(err)); + goto __close; + } + + err = snd_pcm_hw_params_any(handle, hw_params); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_any: %s", snd_strerror(err)); + goto __close; + } + err = snd_pcm_hw_params_set_rate_resample(handle, hw_params, 0); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_rate_resample: %s", snd_strerror(err)); + goto __close; + } + err = snd_pcm_hw_params_set_access(handle, hw_params, access); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_access %s: %s", + snd_pcm_access_name(access), snd_strerror(err)); + goto __close; + } +__format: + err = snd_pcm_hw_params_set_format(handle, hw_params, format); + if (err < 0) { + if (automatic && format == SND_PCM_FORMAT_S16_LE) { + format = SND_PCM_FORMAT_S32_LE; + ksft_print_msg("%s.%d.%d.%d.%s.%s format S16_LE -> S32_LE\n", + cfg_prefix, + data->card, data->device, data->subdevice, + snd_pcm_stream_name(data->stream), + snd_pcm_access_name(access)); + } + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_format %s: %s", + snd_pcm_format_name(format), snd_strerror(err)); + goto __close; + } + err = snd_pcm_hw_params_set_channels(handle, hw_params, channels); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_channels %ld: %s", channels, snd_strerror(err)); + goto __close; + } + rrate = rate; + err = snd_pcm_hw_params_set_rate_near(handle, hw_params, &rrate, 0); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_rate %ld: %s", rate, snd_strerror(err)); + goto __close; + } + if (rrate != rate) { + snprintf(msg, sizeof(msg), "rate mismatch %ld != %ld", rate, rrate); + goto __close; + } + rperiod_size = period_size; + err = snd_pcm_hw_params_set_period_size_near(handle, hw_params, &rperiod_size, 0); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_period_size %ld: %s", period_size, snd_strerror(err)); + goto __close; + } + rbuffer_size = buffer_size; + err = snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &rbuffer_size); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params_set_buffer_size %ld: %s", buffer_size, snd_strerror(err)); + goto __close; + } + err = snd_pcm_hw_params(handle, hw_params); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_hw_params: %s", snd_strerror(err)); + goto __close; + } + + err = snd_pcm_sw_params_current(handle, sw_params); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_sw_params_current: %s", snd_strerror(err)); + goto __close; + } + if (data->stream == SND_PCM_STREAM_PLAYBACK) { + start_threshold = (rbuffer_size / rperiod_size) * rperiod_size; + } else { + start_threshold = rperiod_size; + } + err = snd_pcm_sw_params_set_start_threshold(handle, sw_params, start_threshold); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_sw_params_set_start_threshold %ld: %s", (long)start_threshold, snd_strerror(err)); + goto __close; + } + err = snd_pcm_sw_params_set_avail_min(handle, sw_params, rperiod_size); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_sw_params_set_avail_min %ld: %s", (long)rperiod_size, snd_strerror(err)); + goto __close; + } + err = snd_pcm_sw_params(handle, sw_params); + if (err < 0) { + snprintf(msg, sizeof(msg), "snd_pcm_sw_params: %s", snd_strerror(err)); + goto __close; + } + + ksft_print_msg("%s.%d.%d.%d.%s hw_params.%s.%s.%ld.%ld.%ld.%ld sw_params.%ld\n", + cfg_prefix, + data->card, data->device, data->subdevice, + snd_pcm_stream_name(data->stream), + snd_pcm_access_name(access), + snd_pcm_format_name(format), + (long)rate, (long)channels, + (long)rperiod_size, (long)rbuffer_size, + (long)start_threshold); + + timestamp_now(&tstamp); + for (i = 0; i < 4; i++) { + if (data->stream == SND_PCM_STREAM_PLAYBACK) { + frames = snd_pcm_writei(handle, samples, rate); + if (frames < 0) { + snprintf(msg, sizeof(msg), + "Write failed: expected %d, wrote %li", rate, frames); + goto __close; + } + if (frames < rate) { + snprintf(msg, sizeof(msg), + "expected %d, wrote %li", rate, frames); + goto __close; + } + } else { + frames = snd_pcm_readi(handle, samples, rate); + if (frames < 0) { + snprintf(msg, sizeof(msg), + "expected %d, wrote %li", rate, frames); + goto __close; + } + if (frames < rate) { + snprintf(msg, sizeof(msg), + "expected %d, wrote %li", rate, frames); + goto __close; + } + } + } + + snd_pcm_drain(handle); + ms = timestamp_diff_ms(&tstamp); + if (ms < 3900 || ms > 4100) { + snprintf(msg, sizeof(msg), "time mismatch: expected 4000ms got %lld", ms); + goto __close; + } + + msg[0] = '\0'; + pass = true; +__close: + ksft_test_result(pass, "%s.%d.%d.%d.%s%s%s\n", + cfg_prefix, + data->card, data->device, data->subdevice, + snd_pcm_stream_name(data->stream), + msg[0] ? " " : "", msg); + free(samples); + if (handle) + snd_pcm_close(handle); +} + +#define TESTS_PER_PCM 2 + +int main(void) +{ + struct pcm_data *pcm; + + ksft_print_header(); + + conf_load(); + + find_pcms(); + + ksft_set_plan(num_missing + num_pcms * TESTS_PER_PCM); + + for (pcm = pcm_missing; pcm != NULL; pcm = pcm->next) { + ksft_test_result(false, "test.missing.%d.%d.%d.%s\n", + pcm->card, pcm->device, pcm->subdevice, + snd_pcm_stream_name(pcm->stream)); + } + + for (pcm = pcm_list; pcm != NULL; pcm = pcm->next) { + test_pcm_time1(pcm, "test.time1", "S16_LE", 48000, 2, 512, 4096); + test_pcm_time1(pcm, "test.time2", "S16_LE", 48000, 2, 24000, 192000); + } + + conf_free(); + + ksft_exit_pass(); + + return 0; +} -- 2.35.3

1 0

pcm_old.h does not work with LTO
by GitHub issues - opened 08 Nov '22

08 Nov '22

alsa-project/alsa-lib issue #281 was opened from kkofler: Applications using `ALSA_PCM_OLD_HW_PARAMS_API` and/or `ALSA_PCM_OLD_SW_PARAMS_API` get miscompiled if they are compiled with `-flto` (GCC link-time optimization (LTO)). See https://bugzilla.redhat.com/show_bug.cgi?id=1910437 and in particular https://bugzilla.redhat.com/show_bug.cgi?id=1910437#c23 . The issue is that the `pcm_old.h` header uses constructs like: ```c asm(".symver snd_pcm_hw_params_set_rate_near,snd_pcm_hw_params_set_rate_near(a)ALSA_0.9"); ``` which do not seem to work with LTO enabled. Maybe this needs to use the `symver` attribute instead? In any case, what happens is that the caller in the application attempts to call the new function `snd_pcm_hw_params_set_rate_near(a)ALSA_0.9.0rc4` with the ABI of the old one `snd_pcm_hw_params_set_rate_near(a)ALSA_0.9`, so an integer is passed where ALSA expects a pointer, leading to a segfault (or worse). The workaround is to compile the application without LTO enabled (which makes it work fine), but I believe that this is an upstream ALSA issue and should be fixed in the ALSA header files. Issue URL : https://github.com/alsa-project/alsa-lib/issues/281 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: Correct the header guard of wm8904.h
by Wei Li 08 Nov '22

08 Nov '22

Rename the header guard of wm8904.h from __MFD_WM8994_PDATA_H__ to __WM8904_PDATA_H__ what corresponding with the file name. Fixes: a91eb199e4dc ("ASoC: Initial WM8904 CODEC driver") Signed-off-by: Wei Li <liwei391(a)huawei.com> --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __WM8904_PDATA_H__ +#define __WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.25.1

1 0

[PATCH] ALSA: hda/hdmi - enable runtime pm for more AMD display audio
by Evan Quan 08 Nov '22

08 Nov '22

We are able to power down the GPU and audio via the GPU driver so flag these asics as supporting runtime pm. Signed-off-by: Evan Quan <evan.quan(a)amd.com> Change-Id: Ib7ea9bfe7b62b98afd63f96a2d98795504e7ae04 --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1..e6a33e34efb9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2712,6 +2712,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xab28), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_DEVICE(0x1002, 0xab30), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, { PCI_DEVICE(0x1002, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, -- 2.34.1

2 1

[PATCH 1/2] ALSA: usb-audio: Yet more regression for for the delayed card registration
by Takashi Iwai 08 Nov '22

08 Nov '22

Although we tried to fix the regression for the recent changes with the delayed card registration, it doesn't seem covering the all cases; e.g. on Roland EDIROL M-100FX, where the generic quirk for Roland devices is applied, it misses the card registration because the detection of the last interface (apparently for MIDI) fails. This patch is an attempt to recover from those failures by calling the card register also at the error path for the secondary interfaces. The card register condition is also extended to match with the old check in the previous patch, too (i.e. the simple check of the interface number) for catching the probe with errors. Fixes: 39efc9c8a973 ("ALSA: usb-audio: Fix last interface check for registration") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.suse.com/show_bug.cgi?id=1205111 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 29 ++++++++++++++++++++--------- 1 file changed, 20 insertions(+), 9 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index a5ed11ea1145..26268ffb8274 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -742,6 +742,18 @@ get_alias_quirk(struct usb_device *dev, unsigned int id) return NULL; } +/* register card if we reach to the last interface or to the specified + * one given via option + */ +static int try_to_register_card(struct snd_usb_audio *chip, int ifnum) +{ + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum || + usb_interface_claimed(usb_ifnum_to_if(chip->dev, chip->last_iface))) + return snd_card_register(chip->card); + return 0; +} + /* * probe the active usb device * @@ -880,15 +892,9 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* register card if we reach to the last interface or to the specified - * one given via option - */ - if (check_delayed_register_option(chip) == ifnum || - usb_interface_claimed(usb_ifnum_to_if(dev, chip->last_iface))) { - err = snd_card_register(chip->card); - if (err < 0) - goto __error; - } + err = try_to_register_card(chip, ifnum); + if (err < 0) + goto __error_no_register; if (chip->quirk_flags & QUIRK_FLAG_SHARE_MEDIA_DEVICE) { /* don't want to fail when snd_media_device_create() fails */ @@ -907,6 +913,11 @@ static int usb_audio_probe(struct usb_interface *intf, return 0; __error: + /* in the case of error in secondary interface, still try to register */ + if (chip) + try_to_register_card(chip, ifnum); + + __error_no_register: if (chip) { /* chip->active is inside the chip->card object, * decrement before memory is possibly returned. -- 2.35.3

1 1

[PATCH] ASoC: Trival: Fix spelling error in Kconfig help.
by Maurycy Z 07 Nov '22

07 Nov '22

In a few places in Kconfig files, 'system' is spelled as 'ststem'. This patch changes them to the intended 'system'. Signed-off-by: Maurycy Zalweski <10maurycy10(a)gmail.com> --- sound/soc/fsl/Kconfig | 2 +- sound/soc/generic/Kconfig | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6..c7fe12fbd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -366,7 +366,7 @@ config SND_SOC_IMX_CARD help This option enables audio sound card support for i.MX boards with OF-graph DT bindings. - It also support DPCM of single CPU multi Codec ststem. + It also support DPCM of single CPU multi Codec system. endif # SND_IMX_SOC diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..955ac8d3f 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,7 +7,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support" -- 2.38.1

1 0

[PATCH] ALSA: usb-audio: Add quirks for MacroSilicon MS2100/MS2106 devices (4.9-5.4)
by John Veness 07 Nov '22

07 Nov '22

commit 6e2c9105e0b743c92a157389d40f00b81bdd09fe upstream. Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that the audio stream doesn't sound mono. Also fix initial stream alignment, so that left and right channels are in the correct order. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> Link: https://lore.kernel.org/r/20220624140757.28758-1-john-linux@pelago.org.uk Cc: <stable(a)vger.kernel.org> # 5.4 Cc: <stable(a)vger.kernel.org> # 4.19 Cc: <stable(a)vger.kernel.org> # 4.14 Cc: <stable(a)vger.kernel.org> # 4.9 --- Backported to 4.9-5.4. sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 2 files changed, 59 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c29ccdf9e8bc..5f44b142a8b0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3759,6 +3759,64 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x0021, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 72223545abfd..d61f95dc1abf 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1440,6 +1440,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; -- 2.34.1

2 1

[PATCH] ASoC: SOF: topology: No need to assign core ID if token parsing failed
by Peter Ujfalusi 07 Nov '22

07 Nov '22

Move the return value check before attempting to assign the core ID to the swidget since we are going to fail the sof_widget_ready() and free up swidget anyways. Fixes: 909dadf21aae ("ASoC: SOF: topology: Make DAI widget parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/topology.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 38855dd60617..6a0e7f3b5023 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1344,16 +1344,6 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, break; } - if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { - swidget->core = SOF_DSP_PRIMARY_CORE; - } else { - int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, - swidget->num_tuples); - - if (core >= 0) - swidget->core = core; - } - /* check token parsing reply */ if (ret < 0) { dev_err(scomp->dev, @@ -1365,6 +1355,16 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return ret; } + if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { + swidget->core = SOF_DSP_PRIMARY_CORE; + } else { + int core = sof_get_token_value(SOF_TKN_COMP_CORE_ID, swidget->tuples, + swidget->num_tuples); + + if (core >= 0) + swidget->core = core; + } + /* bind widget to external event */ if (tw->event_type) { if (widget_ops[w->id].bind_event) { -- 2.38.1

2 1

[PATCH v1] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 07 Nov '22

07 Nov '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ------------ v1: cancel delayed work while disabling speaker. --- sound/soc/intel/boards/sof_es8336.c | 60 ++++++++++++++++++++++------- 1 file changed, 46 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..70713e4b07dc 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,46 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) { + cancel_delayed_work(&priv->pcm_pop_work); + gpiod_set_value_cansleep(priv->gpio_speakers, true); + } + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +163,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +373,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +753,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +783,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

2 1

[PATCH v4 0/3] Convert soundwire bindings to DT schema
by Srinivasa Rao Mandadapu 07 Nov '22

07 Nov '22

Convert soundwire bindings text file to DT schema and update device tree entries to follow strict dt-bindings. Changes Since V3: -- Remove subnode description and add appropriate pattern properties. -- Add interrput names in example. -- update some properties description. -- Revert minIteams change in previous version. -- Rebase to latest code base. -- Update dtsi node names, -- Remove redundant property in soundwire node. Srinivasa Rao Mandadapu (3): arm64: dts: qcom: Update soundwire slave node names arm64: dts: qcom: Remove redundant soundwire property dt-bindings: soundwire: Convert text bindings to DT Schema .../devicetree/bindings/soundwire/qcom,sdw.txt | 215 ------------------ .../bindings/soundwire/qcom,soundwire.yaml | 241 +++++++++++++++++++++ arch/arm64/boot/dts/qcom/qrb5165-rb5.dts | 4 +- arch/arm64/boot/dts/qcom/sc7280.dtsi | 1 - arch/arm64/boot/dts/qcom/sdm845-db845c.dts | 4 +- .../boot/dts/qcom/sdm850-lenovo-yoga-c630.dts | 4 +- arch/arm64/boot/dts/qcom/sdm850-samsung-w737.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250-mtp.dts | 4 +- arch/arm64/boot/dts/qcom/sm8250.dtsi | 1 - 9 files changed, 251 insertions(+), 227 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,soundwire.yaml -- 2.7.4

3 9

ALC1220 DualCodec no optical out
by GitHub issues - opened 07 Nov '22

07 Nov '22

alsa-project/alsa-ucm-conf issue #239 was opened from Almoped: With current pipewire and alsa-ucm 1.2.8 I am unable to select profiles which has iec958 output, actually I am unable to select any card profile at all in pavucontrol. This is possibly because all the electrical output jacks are disconnected and optical out does not have jackdetect. Or maybe DualCodecs/HiFi.conf does not configure optical out. If I remove alsa/ucm2/HDA/DualCodecs/HiFi.conf it uses alsa-card-profile/mixer/profile-sets/default.conf, goes on a probing mission I get a working optical out and the possibility to select an array of other card profiles in pavucontrol. Can anyone recommend a better way to solve this? I've attached pactl list cards output with working and nonworking optical out. [pactl_no_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9… [pactl_working_optical.txt](https://github.com/alsa-project/alsa-ucm-conf/fi… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/239 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: dapm: Don't use prefix for regulator name
by Paul Cercueil 07 Nov '22

07 Nov '22

When a component has a prefix, and uses a SND_SOC_DAPM_REGULATOR_SUPPLY, the name of the regulator should not use the prefix, otherwise it won't be properly matched in the DT/ACPI. Fixes: 3caac759681e ("ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling") Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d515e7a78ea8..879cf1be67a9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3645,7 +3645,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, switch (w->id) { case snd_soc_dapm_regulator_supply: - w->regulator = devm_regulator_get(dapm->dev, w->name); + w->regulator = devm_regulator_get(dapm->dev, widget->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); goto request_failed; -- 2.35.1

2 4

[GIT PULL] treewide: timers: Use timer_shutdown*() before freeing timers
by Steven Rostedt 07 Nov '22

07 Nov '22

Linus, As discussed here: https://lore.kernel.org/all/20221106212427.739928660@goodmis.org/ Add a "shutdown" state for timers. This is performed by the new timer_shutdown_sync() and timer_shutdown() function calls. When this is called on a timer, it will no longer be able to be re-armed. This should be called before a timer is freed to prevent it from being re-armed after being removed from the timer queue and then causing a crash in the timer code when the timer triggers. This required renaming some functions that were using the name timer_shutdown() statically to something more appropriate. Then a coccinelle script was executed on the entire kernel tree to find the trivial locations that remove the timer and then frees the object that the timer exists on. These changes are not enough to solve all the locations where timers may be of an issue. But by adding the shutdown infrastructure and the obvious cases, the more complex cases can be added after they have been reviewed more closely. Please pull the following tree, which can be found at: git://git.kernel.org/pub/scm/linux/kernel/git/rostedt/linux-trace.git add-timer-shutdown Tag SHA1: 7685328352dfd2908e23048f563e328dbd3526e9 Head SHA1: 870556da63870e01ade9bb8418ac5a21862f2f10 Steven Rostedt (Google) (5): ARM: spear: Do not use timer namespace for timer_shutdown() function clocksource/drivers/arm_arch_timer: Do not use timer namespace for timer_shutdown() function clocksource/drivers/sp804: Do not use timer namespace for timer_shutdown() function timers: Add timer_shutdown_sync() and timer_shutdown() to be called before freeing timers treewide: Convert del_timer*() to timer_shutdown*() ---- .../RCU/Design/Requirements/Requirements.rst | 2 +- Documentation/core-api/local_ops.rst | 2 +- Documentation/kernel-hacking/locking.rst | 5 ++ arch/arm/mach-spear/time.c | 8 +-- arch/sh/drivers/push-switch.c | 2 +- block/blk-iocost.c | 2 +- block/blk-iolatency.c | 2 +- block/kyber-iosched.c | 2 +- drivers/acpi/apei/ghes.c | 2 +- drivers/atm/idt77252.c | 6 +- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/loop.c | 2 +- drivers/bluetooth/hci_bcsp.c | 2 +- drivers/bluetooth/hci_qca.c | 4 +- drivers/clocksource/arm_arch_timer.c | 12 ++-- drivers/clocksource/timer-sp804.c | 6 +- drivers/gpu/drm/i915/i915_sw_fence.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 +- drivers/input/keyboard/locomokbd.c | 2 +- drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/mouse/alps.c | 2 +- drivers/isdn/mISDN/l1oip_core.c | 4 +- drivers/isdn/mISDN/timerdev.c | 4 +- drivers/leds/trigger/ledtrig-activity.c | 2 +- drivers/leds/trigger/ledtrig-heartbeat.c | 2 +- drivers/leds/trigger/ledtrig-pattern.c | 2 +- drivers/leds/trigger/ledtrig-transient.c | 2 +- drivers/media/pci/ivtv/ivtv-driver.c | 2 +- drivers/media/usb/pvrusb2/pvrusb2-hdw.c | 16 +++--- drivers/media/usb/s2255/s2255drv.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 6 +- drivers/net/ethernet/marvell/sky2.c | 2 +- drivers/net/ethernet/sun/sunvnet.c | 2 +- drivers/net/usb/sierra_net.c | 2 +- .../wireless/broadcom/brcm80211/brcmfmac/btcoex.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-dbg-tlv.c | 2 +- drivers/net/wireless/intel/iwlwifi/mvm/sta.c | 2 +- drivers/net/wireless/intersil/hostap/hostap_ap.c | 2 +- drivers/net/wireless/marvell/mwifiex/main.c | 2 +- drivers/net/wireless/microchip/wilc1000/hif.c | 6 +- drivers/nfc/pn533/pn533.c | 2 +- drivers/nfc/pn533/uart.c | 2 +- drivers/pcmcia/bcm63xx_pcmcia.c | 2 +- drivers/pcmcia/electra_cf.c | 2 +- drivers/pcmcia/omap_cf.c | 2 +- drivers/pcmcia/pd6729.c | 4 +- drivers/pcmcia/yenta_socket.c | 4 +- drivers/scsi/qla2xxx/qla_edif.c | 4 +- drivers/staging/media/atomisp/i2c/atomisp-lm3554.c | 2 +- drivers/tty/n_gsm.c | 2 +- drivers/tty/sysrq.c | 2 +- drivers/usb/gadget/udc/m66592-udc.c | 2 +- drivers/usb/serial/garmin_gps.c | 2 +- drivers/usb/serial/mos7840.c | 4 +- fs/ext4/super.c | 2 +- fs/nilfs2/segment.c | 2 +- include/linux/timer.h | 62 +++++++++++++++++++-- kernel/time/timer.c | 64 ++++++++++++---------- net/802/garp.c | 2 +- net/802/mrp.c | 4 +- net/bridge/br_multicast.c | 8 +-- net/bridge/br_multicast_eht.c | 4 +- net/core/gen_estimator.c | 2 +- net/ipv4/ipmr.c | 2 +- net/ipv6/ip6mr.c | 2 +- net/mac80211/mesh_pathtbl.c | 2 +- net/netfilter/ipset/ip_set_list_set.c | 2 +- net/netfilter/ipvs/ip_vs_lblc.c | 2 +- net/netfilter/ipvs/ip_vs_lblcr.c | 2 +- net/netfilter/xt_IDLETIMER.c | 4 +- net/netfilter/xt_LED.c | 2 +- net/rxrpc/conn_object.c | 2 +- net/sched/cls_flow.c | 2 +- net/sunrpc/svc.c | 2 +- net/tipc/discover.c | 2 +- net/tipc/monitor.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 78 files changed, 207 insertions(+), 148 deletions(-) ---------------------------

2 1

[PATCH] ASoC: wm8962: Wait for updated value of WM8962_CLOCKING1 register
by Chancel Liu 07 Nov '22

07 Nov '22

DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate correct frequency of LRCLK and BCLK. Sometimes the read-only value can't be updated timely after enabling SYSCLK. This results in wrong calculation values. Delay is introduced here to wait for newest value from register. The time of the delay should be at least 500~1000us according to test. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b4b4355c6728..563843565f68 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2503,6 +2503,14 @@ static void wm8962_configure_bclk(struct snd_soc_component *component) snd_soc_component_update_bits(component, WM8962_CLOCKING2, WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + /* DSPCLK_DIV field in WM8962_CLOCKING1 register is used to generate + * correct frequency of LRCLK and BCLK. Sometimes the read-only value + * can't be updated timely after enabling SYSCLK. This results in wrong + * calculation values. Delay is introduced here to wait for newest + * value from register. The time of the delay should be at least + * 500~1000us according to test. + */ + msleep(1); dspclk = snd_soc_component_read(component, WM8962_CLOCKING1); if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_ON) -- 2.25.1

2 1

Question regarding delayed trigger in dpcm_set_fe_update_state()
by Cezary Rojewski 07 Nov '22

07 Nov '22

Hello, Writing with question regarding dpcm_set_fe_update_state() function which is part of sound/soc/soc-pcm.c file and has been introduced with commit "ASoC: dpcm: Fix race between FE/BE updates and trigger" [1]. The part that concerns me is the invocation of dpcm_fe_dai_do_trigger() regardless of the actual state given DPCM is in (actual state, not the DPCM_UPDATE_XX). The conditional invocation of said _trigger() and addition of .trigger_pending field are here to address a race where PCM state is being modified from multiple locations simultaneously, at least judging by the commit's description. Note that the dpcm_set_fe_update_state() is called from all the dai-ops i.e.: startup, shutdown, hw_params, prepare and hw_free. Now, given that knowledge, we could end up in scenario where dpcm_fe_dai_do_trigger() is invoked as a part of dpcm_fe_dai_hw_free(). dpcm_set_fe_update_state() is called there twice, once with DPCM_UPDATE_FE and once with DPCM_UPDATE_NO. The second case is the more interesting one since it's called **after** ->hw_free() callback is invoked for all the DAIs. dpcm_fe_dai_hw_free() dpcm_set_fe_update_state(DPCM_UPDATE_FE) // fine (...) dpcm_be_dai_hw_free() // data allocated in hw_params // is freed here (...) dpcm_set_fe_update_state(DPCM_UPDATE_NO) // not fine The last is *not fine* if the .trigger_pending is not a zero, and can lead to panic as code used during ->trigger() is often manipulating data allocated during ->hw_params() but that data has just been freed with ->hw_free(). Is what I'm looking at a bug? Or, perhaps there's something I'm missing in the picture. From my study, it seems that only dpcm_fe_dai_prepare() is a safe place for calling dpcm_fe_dai_do_trigger() - once .trigger_pending is taken into account. Any input is much appreciated. [1]: https://lore.kernel.org/all/1415116348-11792-1-git-send-email-tiwai@suse.de… Regards, Czarek

2 3

回复: Re: 回复: Re: 回复: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue
by 艾超 07 Nov '22

07 Nov '22

I tested the hotplug ãrecord and catch the picture, this interface reset doesn't cause any problem. In snd_usb_endpoint_configure, It set false to interface to deselect altsetting at first. But , if skip the interface clear (ex. ep->cur_rate = 0;), there is no sound output. ---- ä¸»ãé¢ï¼Re: åå¤: Re: åå¤: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue æ¥ãæï¼2022-11-07 16:15 åä»¶äººï¼Takashi Iwai æ¶ä»¶äººï¼è¾è¶ On Mon, 07 Nov 2022 02:33:57 +0100, è¾è¶ wrote: > > Yes, the audio device is a UAC1 device, and it used set_sample_rate_v1 to > set sample. > > So, I add a quirk to keep up the USB interface up even if the audio stream > is closed. Well, my question is also about the parameter changes. This interface reset doesn't cause any problem? Or we should skip the interface clear there, too? Takashi > > ---- > > ä¸»ãé¢ï¼Re: åå¤: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero > disconnect issue > æ¥ãæï¼2022-11-04 17:59 > åä»¶äººï¼Takashi Iwai > æ¶ä»¶äººï¼è¾è¶ > > On Fri, 04 Nov 2022 10:50:48 +0100, > è¾è¶ wrote: > > > > I 'm sorry, I don't know whether you receiver my mail for 'lsusb -v' to > > Hamedal C20 > > I received your previous mails including lsusb -v output. But the > post was filtered out by some reason? > > So the audio device looks like a UAC1 device. And this one requires > to keep up the USB interface up even if the audio stream is closed > (not about the parameter changes)? > > Takashi > >

1 0

[PATCH] ALSA: usb-audio: Add quirks for MacroSilicon MS2100/MS2106 devices (5.10)
by John Veness 07 Nov '22

07 Nov '22

[ Upstream commit 6e2c9105e0b743c92a157389d40f00b81bdd09fe] Backported to 5.10. Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that the audio stream doesn't sound mono. Also fix initial stream alignment, so that left and right channels are in the correct order. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> Cc: <stable(a)vger.kernel.org> # 5.10 --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 1 + 2 files changed, 53 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1ac91c46da3c..a51591f68ae6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3656,6 +3656,58 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 41f5d8242478..04a691bc560c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1508,6 +1508,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; -- 2.34.1

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[PATCH] ALSA: cs5535audio: Remove the redundant assignment
by Jiapeng Chong 07 Nov '22

07 Nov '22

Variable 'desc_addr' set but not used. sound/pci/cs5535audio/cs5535audio_pcm.c:113:12: warning: variable 'desc_addr' set but not used. Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=2739 Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 0db24cc4d916..9c88e99e3750 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -110,7 +110,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, unsigned int period_bytes) { unsigned int i; - u32 addr, desc_addr, jmpprd_addr; + u32 addr, jmpprd_addr; struct cs5535audio_dma_desc *lastdesc; if (periods > CS5535AUDIO_MAX_DESCRIPTORS) @@ -131,14 +131,12 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, /* the u32 cast is okay because in snd*create we successfully told pci alloc that we're only 32 bit capable so the upper will be 0 */ addr = (u32) substream->runtime->dma_addr; - desc_addr = (u32) dma->desc_buf.addr; for (i = 0; i < periods; i++) { struct cs5535audio_dma_desc *desc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); desc->size = cpu_to_le16(period_bytes); desc->ctlreserved = cpu_to_le16(PRD_EOP); - desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes; } /* we reserved one dummy descriptor at the end to do the PRD jump */ -- 2.20.1.7.g153144c

2 1

回复: Re: 回复: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue
by 艾超 07 Nov '22

07 Nov '22

Yes, the audio device is a UAC1 device, and it used set_sample_rate_v1 to set sample. So, I add a quirk to keep up the USB interface up even if the audio stream is closed. ---- ä¸»ãé¢ï¼Re: åå¤: Re: [PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue æ¥ãæï¼2022-11-04 17:59 åä»¶äººï¼Takashi Iwai æ¶ä»¶äººï¼è¾è¶ On Fri, 04 Nov 2022 10:50:48 +0100, è¾è¶ wrote: > > I 'm sorry, I don't know whether you receiver my mail for 'lsusb -v' to > Hamedal C20 I received your previous mails including lsusb -v output. But the post was filtered out by some reason? So the audio device looks like a UAC1 device. And this one requires to keep up the USB interface up even if the audio stream is closed (not about the parameter changes)? Takashi

1 0

[PATCH v6a 0/5] timers: Use timer_shutdown*() before freeing timers
by Steven Rostedt 06 Nov '22

06 Nov '22

del_timer_sync() is often called before the object that owns the timer is freed. But sometimes there's a race that enables the timer again before it is freed and causes a use after free when that timer triggers. This patch set adds a new "shutdown" timer state, which is set on the new timer_shutdown() API. Once a timer is in this state, it can not be re-armed and if it is, it will warn. The first three patches change existing timer_shutdown() functions used locally in ARM and some drivers to better namespace names. The fourth patch implements the new API. The fifth patch is now a treewide patch that uses a coccinelle script to convert the trivial locations where a del_timer*() is called on a timer of an object that is freed immediately afterward (or at least in the same function). Changes since v5a: https://lore.kernel.org/all/20221106054535.709068702@goodmis.org/ - Updated the script to make ptr and slab into expressions instead of using identifiers (Julia Lawall and Linus Torvalds) Steven Rostedt (Google) (5): ARM: spear: Do not use timer namespace for timer_shutdown() function clocksource/drivers/arm_arch_timer: Do not use timer namespace for timer_shutdown() function clocksource/drivers/sp804: Do not use timer namespace for timer_shutdown() function timers: Add timer_shutdown_sync() and timer_shutdown() to be called before freeing timers treewide: Convert del_timer*() to timer_shutdown*() ---- .../RCU/Design/Requirements/Requirements.rst | 2 +- Documentation/core-api/local_ops.rst | 2 +- Documentation/kernel-hacking/locking.rst | 5 ++ arch/arm/mach-spear/time.c | 8 +-- arch/sh/drivers/push-switch.c | 2 +- block/blk-iocost.c | 2 +- block/blk-iolatency.c | 2 +- block/kyber-iosched.c | 2 +- drivers/acpi/apei/ghes.c | 2 +- drivers/atm/idt77252.c | 6 +- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/loop.c | 2 +- drivers/bluetooth/hci_bcsp.c | 2 +- drivers/bluetooth/hci_qca.c | 4 +- drivers/clocksource/arm_arch_timer.c | 12 ++-- drivers/clocksource/timer-sp804.c | 6 +- drivers/gpu/drm/i915/i915_sw_fence.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 +- drivers/input/keyboard/locomokbd.c | 2 +- drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/mouse/alps.c | 2 +- drivers/isdn/mISDN/l1oip_core.c | 4 +- drivers/isdn/mISDN/timerdev.c | 4 +- drivers/leds/trigger/ledtrig-activity.c | 2 +- drivers/leds/trigger/ledtrig-heartbeat.c | 2 +- drivers/leds/trigger/ledtrig-pattern.c | 2 +- drivers/leds/trigger/ledtrig-transient.c | 2 +- drivers/media/pci/ivtv/ivtv-driver.c | 2 +- drivers/media/usb/pvrusb2/pvrusb2-hdw.c | 16 +++--- drivers/media/usb/s2255/s2255drv.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 6 +- drivers/net/ethernet/marvell/sky2.c | 2 +- drivers/net/ethernet/sun/sunvnet.c | 2 +- drivers/net/usb/sierra_net.c | 2 +- .../wireless/broadcom/brcm80211/brcmfmac/btcoex.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-dbg-tlv.c | 2 +- drivers/net/wireless/intel/iwlwifi/mvm/sta.c | 2 +- drivers/net/wireless/intersil/hostap/hostap_ap.c | 2 +- drivers/net/wireless/marvell/mwifiex/main.c | 2 +- drivers/net/wireless/microchip/wilc1000/hif.c | 6 +- drivers/nfc/pn533/pn533.c | 2 +- drivers/nfc/pn533/uart.c | 2 +- drivers/pcmcia/bcm63xx_pcmcia.c | 2 +- drivers/pcmcia/electra_cf.c | 2 +- drivers/pcmcia/omap_cf.c | 2 +- drivers/pcmcia/pd6729.c | 4 +- drivers/pcmcia/yenta_socket.c | 4 +- drivers/scsi/qla2xxx/qla_edif.c | 4 +- drivers/staging/media/atomisp/i2c/atomisp-lm3554.c | 2 +- drivers/tty/n_gsm.c | 2 +- drivers/tty/sysrq.c | 2 +- drivers/usb/gadget/udc/m66592-udc.c | 2 +- drivers/usb/serial/garmin_gps.c | 2 +- drivers/usb/serial/mos7840.c | 4 +- fs/ext4/super.c | 2 +- fs/nilfs2/segment.c | 2 +- include/linux/timer.h | 62 +++++++++++++++++++-- kernel/time/timer.c | 64 ++++++++++++---------- net/802/garp.c | 2 +- net/802/mrp.c | 4 +- net/bridge/br_multicast.c | 8 +-- net/bridge/br_multicast_eht.c | 4 +- net/core/gen_estimator.c | 2 +- net/ipv4/ipmr.c | 2 +- net/ipv6/ip6mr.c | 2 +- net/mac80211/mesh_pathtbl.c | 2 +- net/netfilter/ipset/ip_set_list_set.c | 2 +- net/netfilter/ipvs/ip_vs_lblc.c | 2 +- net/netfilter/ipvs/ip_vs_lblcr.c | 2 +- net/netfilter/xt_IDLETIMER.c | 4 +- net/netfilter/xt_LED.c | 2 +- net/rxrpc/conn_object.c | 2 +- net/sched/cls_flow.c | 2 +- net/sunrpc/svc.c | 2 +- net/tipc/discover.c | 2 +- net/tipc/monitor.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 78 files changed, 207 insertions(+), 148 deletions(-)

1 0

[PATCH AUTOSEL 5.4 01/12] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit de71d7567e358effd06dfc3e2a154b25f1331c10 ] This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c5667b149c70..d6d4b4121369 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2084,6 +2084,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2116,9 +2119,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

2 8

[PATCH AUTOSEL 4.19 1/7] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit de71d7567e358effd06dfc3e2a154b25f1331c10 ] This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 9b33c87c2fe4..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2085,6 +2085,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2117,9 +2120,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 2

[PATCH AUTOSEL 5.10 01/16] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit de71d7567e358effd06dfc3e2a154b25f1331c10 ] This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b7f5e5391fdb..2ed3fa67027d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2083,6 +2083,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2115,9 +2118,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 10

[PATCH AUTOSEL 5.15 01/18] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit de71d7567e358effd06dfc3e2a154b25f1331c10 ] This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c8adce8936bc..621598608bf0 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2087,6 +2087,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2119,9 +2122,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 11

[PATCH AUTOSEL 6.0 23/30] ASoC: amd: yc: Add Lenovo Thinkbook 14+ 2022 21D0 to quirks table
by Sasha Levin 06 Nov '22

06 Nov '22

From: Leohearts <leohearts(a)leohearts.com> [ Upstream commit a75481fa00cc06a8763e1795b93140407948c03a ] Lenovo Thinkbook 14+ 2022 (ThinkBook 14 G4+ ARA) uses Ryzen 6000 processor, and has the same microphone problem as other ThinkPads with AMD Ryzen 6000 series CPUs, which has been listed in https://bugzilla.kernel.org/show_bug.cgi?id=216267. Adding 21D0 to quirks table solves this microphone problem for ThinkBook 14 G4+ ARA. Signed-off-by: Taroe Leohearts <leohearts(a)leohearts.com> Link: https://lore.kernel.org/r/26B141B486BEF706+313d1732-e00c-ea41-3123-0d048d40… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 09a8aceff22f..6c0f1de10429 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -52,6 +52,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 22/30] ASoC: amd: yc: Adding Lenovo ThinkBook 14 Gen 4+ ARA and Lenovo ThinkBook 16 Gen 4+ ARA to the Quirks List
by Sasha Levin 06 Nov '22

06 Nov '22

From: linkt <xazrael(a)hotmail.com> [ Upstream commit a450b5c8739248069e11f72129fca61a56125577 ] Lenovo ThinkBook 14 Gen 4+ ARA and ThinkBook 16 Gen 4+ ARA need to be added to the list of quirks for the microphone to work properly. Signed-off-by: linkt <xazrael(a)hotmail.com> Reviewed-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://lore.kernel.org/r/MEYPR01MB8397A3C27DE6206FA3EF834DB6239@MEYPR01MB8… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 2cb50d5cf1a9..09a8aceff22f 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,6 +45,20 @@ static struct snd_soc_card acp6x_card = { }; static const struct dmi_system_id yc_acp_quirk_table[] = { + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D0"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21D1"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 21/30] ASoC: rt1308-sdw: add the default value of some registers
by Sasha Levin 06 Nov '22

06 Nov '22

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 75d8b1662ca5c20cf8365575222abaef18ff1f50 ] The driver missed the default value of register 0xc070/0xc360. This patch adds that default value to avoid invalid register access when the device doesn't be enumerated yet. BugLink: https://github.com/thesofproject/linux/issues/3924 Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20221019095715.31082-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1308-sdw.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 6668e19d85d4..b5f231f708cb 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -139,10 +139,12 @@ static const struct reg_default rt1308_reg_defaults[] = { { 0x3005, 0x23 }, { 0x3008, 0x02 }, { 0x300a, 0x00 }, + { 0xc000 | (RT1308_DATA_PATH << 4), 0x00 }, { 0xc003 | (RT1308_DAC_SET << 4), 0x00 }, { 0xc000 | (RT1308_POWER << 4), 0x00 }, { 0xc001 | (RT1308_POWER << 4), 0x00 }, { 0xc002 | (RT1308_POWER << 4), 0x00 }, + { 0xc000 | (RT1308_POWER_STATUS << 4), 0x00 }, }; #define RT1308_SDW_OFFSET 0xc000 -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 19/30] ASoC: Intel: sof_rt5682: Add quirk for Rex board
by Sasha Levin 06 Nov '22

06 Nov '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit b4dd2e3758709aa8a2abd1ac34c56bd09b980039 ] Add mtl_mx98357_rt5682 driver data for Chrome Rex board support. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221017205728.210813-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312245..30c53dca342e 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -225,6 +225,18 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(2) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(0) | + SOF_RT5682_NUM_HDMIDEV(4) + ), + }, {} }; -- 2.35.1

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[PATCH AUTOSEL 6.0 15/30] ASoC: Intel: sof_sdw: add quirk variant for LAPBC710 NUC15
by Sasha Levin 06 Nov '22

06 Nov '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 41deb2db64997d01110faaf763bd911d490dfde7 ] Some NUC15 LAPBC710 devices don't expose the same DMI information as the Intel reference, add additional entry in the match table. BugLink: https://github.com/thesofproject/linux/issues/3885 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20221017204054.207512-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2ff30b40a1e4..ee9857dc3135 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -202,6 +202,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD1), }, + { + /* NUC15 LAPBC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPBC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD1), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.35.1

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[PATCH AUTOSEL 6.0 14/30] ASoC: codecs: jz4725b: fix capture selector naming
by Sasha Levin 06 Nov '22

06 Nov '22

From: Siarhei Volkau <lis8215(a)gmail.com> [ Upstream commit 80852f8268769715db335a22305e81a0c4a38a84 ] At the moment Capture source selector appears on Playback tab in the alsamixer and has a senseless name. Let's fix that. Signed-off-by: Siarhei Volkau <lis8215(a)gmail.com> Link: https://lore.kernel.org/r/20221016132648.3011729-5-lis8215@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/jz4725b.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 4363d898a7d4..d57c2c6a3add 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -183,7 +183,7 @@ static SOC_VALUE_ENUM_SINGLE_DECL(jz4725b_codec_adc_src_enum, jz4725b_codec_adc_src_texts, jz4725b_codec_adc_src_values); static const struct snd_kcontrol_new jz4725b_codec_adc_src_ctrl = - SOC_DAPM_ENUM("Route", jz4725b_codec_adc_src_enum); + SOC_DAPM_ENUM("ADC Source Capture Route", jz4725b_codec_adc_src_enum); static const struct snd_kcontrol_new jz4725b_codec_mixer_controls[] = { SOC_DAPM_SINGLE("Line In Bypass", JZ4725B_CODEC_REG_CR1, @@ -228,7 +228,7 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "Capture", JZ4725B_CODEC_REG_PMR1, REG_PMR1_SB_ADC_OFFSET, 1), - SND_SOC_DAPM_MUX("ADC Source", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_MUX("ADC Source Capture Route", SND_SOC_NOPM, 0, 0, &jz4725b_codec_adc_src_ctrl), /* Mixer */ @@ -287,11 +287,11 @@ static const struct snd_soc_dapm_route jz4725b_codec_dapm_routes[] = { {"Mixer", NULL, "DAC to Mixer"}, {"Mixer to ADC", NULL, "Mixer"}, - {"ADC Source", "Mixer", "Mixer to ADC"}, - {"ADC Source", "Line In", "Line In"}, - {"ADC Source", "Mic 1", "Mic 1"}, - {"ADC Source", "Mic 2", "Mic 2"}, - {"ADC", NULL, "ADC Source"}, + {"ADC Source Capture Route", "Mixer", "Mixer to ADC"}, + {"ADC Sourc Capture Routee", "Line In", "Line In"}, + {"ADC Source Capture Route", "Mic 1", "Mic 1"}, + {"ADC Source Capture Route", "Mic 2", "Mic 2"}, + {"ADC", NULL, "ADC Source Capture Route"}, {"Out Stage", NULL, "Mixer"}, {"HP Out", NULL, "Out Stage"}, -- 2.35.1

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[PATCH AUTOSEL 6.0 13/30] ASoC: codecs: jz4725b: use right control for Capture Volume
by Sasha Levin 06 Nov '22

06 Nov '22

From: Siarhei Volkau <lis8215(a)gmail.com> [ Upstream commit 1538e2c8c9b7e7a656effcc6e4e7cfe8c1b405fd ] Line In Bypass control is used as Master Capture at the moment this is completely incorrect. Current control routed to Mixer instead of ADC, thus can't affect Capture path. ADC control shall be used instead. ADC volume control parameters are different, so the patch fixes that as well. Manual says (16.6.3.2 Programmable input attenuation amplifier: PGATM) that gain varies in range 0dB..22.5dB with 1.5dB step. Signed-off-by: Siarhei Volkau <lis8215(a)gmail.com> Link: https://lore.kernel.org/r/20221016132648.3011729-4-lis8215@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/jz4725b.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 72549ee2e789..4363d898a7d4 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -136,13 +136,16 @@ enum { #define REG_CGR3_GO1L_OFFSET 0 #define REG_CGR3_GO1L_MASK (0x1f << REG_CGR3_GO1L_OFFSET) +#define REG_CGR10_GIL_OFFSET 0 +#define REG_CGR10_GIR_OFFSET 4 + struct jz_icdc { struct regmap *regmap; void __iomem *base; struct clk *clk; }; -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_adc_tlv, 0, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_dac_tlv, -2250, 150, 0); static const struct snd_kcontrol_new jz4725b_codec_controls[] = { @@ -151,11 +154,11 @@ static const struct snd_kcontrol_new jz4725b_codec_controls[] = { REG_CGR1_GODL_OFFSET, REG_CGR1_GODR_OFFSET, 0xf, 1, jz4725b_dac_tlv), - SOC_DOUBLE_R_TLV("Master Capture Volume", - JZ4725B_CODEC_REG_CGR3, - JZ4725B_CODEC_REG_CGR2, - REG_CGR2_GO1R_OFFSET, - 0x1f, 1, jz4725b_line_tlv), + SOC_DOUBLE_TLV("Master Capture Volume", + JZ4725B_CODEC_REG_CGR10, + REG_CGR10_GIL_OFFSET, + REG_CGR10_GIR_OFFSET, + 0xf, 0, jz4725b_adc_tlv), SOC_SINGLE("Master Playback Switch", JZ4725B_CODEC_REG_CR1, REG_CR1_DAC_MUTE_OFFSET, 1, 1), -- 2.35.1

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[PATCH AUTOSEL 6.0 12/30] ASoC: codecs: jz4725b: fix reported volume for Master ctl
by Sasha Levin 06 Nov '22

06 Nov '22

From: Siarhei Volkau <lis8215(a)gmail.com> [ Upstream commit 088777bf65b98cfa4b5378119d0a7d49a58ece44 ] DAC volume control is the Master Playback Volume at the moment and it reports wrong levels in alsamixer and other alsa apps. The patch fixes that, as stated in manual on the jz4725b SoC (16.6.3.4 Programmable attenuation: GOD) the ctl range varies from -22.5dB to 0dB with 1.5dB step. Signed-off-by: Siarhei Volkau <lis8215(a)gmail.com> Link: https://lore.kernel.org/r/20221016132648.3011729-3-lis8215@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/jz4725b.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index cc7a48c96aa4..72549ee2e789 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -142,8 +142,8 @@ struct jz_icdc { struct clk *clk; }; -static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_dac_tlv, -2250, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_LINEAR(jz4725b_line_tlv, -1500, 600); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(jz4725b_dac_tlv, -2250, 150, 0); static const struct snd_kcontrol_new jz4725b_codec_controls[] = { SOC_DOUBLE_TLV("Master Playback Volume", -- 2.35.1

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[PATCH AUTOSEL 6.0 11/30] ASoC: codecs: jz4725b: add missed Line In power control bit
by Sasha Levin 06 Nov '22

06 Nov '22

From: Siarhei Volkau <lis8215(a)gmail.com> [ Upstream commit 1013999b431b4bcdc1f5ae47dd3338122751db31 ] Line In path stayed powered off during capturing or bypass to mixer. Signed-off-by: Siarhei Volkau <lis8215(a)gmail.com> Link: https://lore.kernel.org/r/20221016132648.3011729-2-lis8215@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/jz4725b.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 5201a8f6d7b6..cc7a48c96aa4 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -236,7 +236,8 @@ static const struct snd_soc_dapm_widget jz4725b_codec_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DAC to Mixer", JZ4725B_CODEC_REG_CR1, REG_CR1_DACSEL_OFFSET, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Line In", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Line In", JZ4725B_CODEC_REG_PMR1, + REG_PMR1_SB_LIN_OFFSET, 1, NULL, 0), SND_SOC_DAPM_MIXER("HP Out", JZ4725B_CODEC_REG_CR1, REG_CR1_HP_DIS_OFFSET, 1, NULL, 0), -- 2.35.1

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[PATCH AUTOSEL 6.0 09/30] ASoC: wm8962: Add an event handler for TEMP_HP and TEMP_SPK
by Sasha Levin 06 Nov '22

06 Nov '22

From: Xiaolei Wang <xiaolei.wang(a)windriver.com> [ Upstream commit ee1aa2ae3eaa96e70229fa61deee87ef4528ffdf ] In wm8962 driver, the WM8962_ADDITIONAL_CONTROL_4 is used as a volatile register, but this register mixes a bunch of volatile status bits and a bunch of non-volatile control bits. The dapm widgets TEMP_HP and TEMP_SPK leverages the control bits in this register. After the wm8962 probe, the regmap will bet set to cache only mode, then a read error like below would be triggered when trying to read the initial power state of the dapm widgets TEMP_HP and TEMP_SPK. wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 In order to fix this issue, we add event handler to actually power up/down these widgets. With this change, we also need to explicitly power off these widgets in the wm8962 probe since they are enabled by default. Signed-off-by: Xiaolei Wang <xiaolei.wang(a)windriver.com> Tested-by: Adam Ford <aford173(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010092014.2229246-1-xiaolei.wang@windriver.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8962.c | 54 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 398c448ea854..6df06fba4377 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1840,6 +1840,49 @@ SOC_SINGLE_TLV("SPKOUTR Mixer DACR Volume", WM8962_SPEAKER_MIXER_5, 4, 1, 0, inmix_tlv), }; +static int tp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + int ret, reg, val, mask; + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + dev_err(component->dev, "Failed to resume device: %d\n", ret); + return ret; + } + + reg = WM8962_ADDITIONAL_CONTROL_4; + + if (!strcmp(w->name, "TEMP_HP")) { + mask = WM8962_TEMP_ENA_HP_MASK; + val = WM8962_TEMP_ENA_HP; + } else if (!strcmp(w->name, "TEMP_SPK")) { + mask = WM8962_TEMP_ENA_SPK_MASK; + val = WM8962_TEMP_ENA_SPK; + } else { + pm_runtime_put(component->dev); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + val = 0; + fallthrough; + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, reg, mask, val); + break; + default: + WARN(1, "Invalid event %d\n", event); + pm_runtime_put(component->dev); + return -EINVAL; + } + + pm_runtime_put(component->dev); + + return 0; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2140,8 +2183,10 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_HP", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", SND_SOC_NOPM, 0, 0, tp_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -3763,6 +3808,11 @@ static int wm8962_i2c_probe(struct i2c_client *i2c) if (ret < 0) goto err_pm_runtime; + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_HP_MASK, 0); + regmap_update_bits(wm8962->regmap, WM8962_ADDITIONAL_CONTROL_4, + WM8962_TEMP_ENA_SPK_MASK, 0); + regcache_cache_only(wm8962->regmap, true); /* The drivers should power up as needed */ -- 2.35.1

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[PATCH AUTOSEL 6.0 08/30] ASoC: rt1019: Fix the TDM settings
by Sasha Levin 06 Nov '22

06 Nov '22

From: Derek Fang <derek.fang(a)realtek.com> [ Upstream commit f2635d45a750182c6d5de15e2d6b059e0c302d7e ] Complete the missing and correct the TDM settings. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/20221012030102.4042-1-derek.fang@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1019.c | 20 +++++++++++--------- sound/soc/codecs/rt1019.h | 6 ++++++ 2 files changed, 17 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index b66bfecbb879..49f527c61a7a 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -391,18 +391,18 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; - unsigned int val = 0, rx_slotnum; + unsigned int cn = 0, cl = 0, rx_slotnum; int ret = 0, first_bit; switch (slots) { case 4: - val |= RT1019_I2S_TX_4CH; + cn = RT1019_I2S_TX_4CH; break; case 6: - val |= RT1019_I2S_TX_6CH; + cn = RT1019_I2S_TX_6CH; break; case 8: - val |= RT1019_I2S_TX_8CH; + cn = RT1019_I2S_TX_8CH; break; case 2: break; @@ -412,16 +412,16 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (slot_width) { case 20: - val |= RT1019_I2S_DL_20; + cl = RT1019_TDM_CL_20; break; case 24: - val |= RT1019_I2S_DL_24; + cl = RT1019_TDM_CL_24; break; case 32: - val |= RT1019_I2S_DL_32; + cl = RT1019_TDM_CL_32; break; case 8: - val |= RT1019_I2S_DL_8; + cl = RT1019_TDM_CL_8; break; case 16: break; @@ -470,8 +470,10 @@ static int rt1019_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, goto _set_tdm_err_; } + snd_soc_component_update_bits(component, RT1019_TDM_1, + RT1019_TDM_CL_MASK, cl); snd_soc_component_update_bits(component, RT1019_TDM_2, - RT1019_I2S_CH_TX_MASK | RT1019_I2S_DF_MASK, val); + RT1019_I2S_CH_TX_MASK, cn); _set_tdm_err_: return ret; diff --git a/sound/soc/codecs/rt1019.h b/sound/soc/codecs/rt1019.h index 64df831eeb72..48ba15efb48d 100644 --- a/sound/soc/codecs/rt1019.h +++ b/sound/soc/codecs/rt1019.h @@ -95,6 +95,12 @@ #define RT1019_TDM_BCLK_MASK (0x1 << 6) #define RT1019_TDM_BCLK_NORM (0x0 << 6) #define RT1019_TDM_BCLK_INV (0x1 << 6) +#define RT1019_TDM_CL_MASK (0x7) +#define RT1019_TDM_CL_8 (0x4) +#define RT1019_TDM_CL_32 (0x3) +#define RT1019_TDM_CL_24 (0x2) +#define RT1019_TDM_CL_20 (0x1) +#define RT1019_TDM_CL_16 (0x0) /* 0x0401 TDM Control-2 */ #define RT1019_I2S_CH_TX_MASK (0x3 << 6) -- 2.35.1

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[PATCH AUTOSEL 6.0 07/30] ASoC: rt5682s: Fix the TDM Tx settings
by Sasha Levin 06 Nov '22

06 Nov '22

From: Derek Fang <derek.fang(a)realtek.com> [ Upstream commit d94bf16e920047c9b4ea2b57f7b53b4ff5039d9f ] Complete the missing and correct the TDM Tx settings. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/20221012031320.6980-1-derek.fang@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682s.c | 15 +++++++++++++-- sound/soc/codecs/rt5682s.h | 1 + 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index eb47e7cd485a..95fe993d59cb 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1932,7 +1932,7 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; - unsigned int cl, val = 0; + unsigned int cl, val = 0, tx_slotnum; if (tx_mask || rx_mask) snd_soc_component_update_bits(component, @@ -1941,6 +1941,16 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, snd_soc_component_update_bits(component, RT5682S_TDM_ADDA_CTRL_2, RT5682S_TDM_EN, 0); + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + if (tx_slotnum > slots) { + dev_err(component->dev, "Invalid or oversized Tx slots.\n"); + return -EINVAL; + } + val |= (tx_slotnum - 1) << RT5682S_TDM_ADC_DL_SFT; + } + switch (slots) { case 4: val |= RT5682S_TDM_TX_CH_4; @@ -1961,7 +1971,8 @@ static int rt5682s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, } snd_soc_component_update_bits(component, RT5682S_TDM_CTRL, - RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK, val); + RT5682S_TDM_TX_CH_MASK | RT5682S_TDM_RX_CH_MASK | + RT5682S_TDM_ADC_DL_MASK, val); switch (slot_width) { case 8: diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 7353831c73dd..b660a311b6c2 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -899,6 +899,7 @@ #define RT5682S_TDM_RX_CH_8 (0x3 << 8) #define RT5682S_TDM_ADC_LCA_MASK (0x7 << 4) #define RT5682S_TDM_ADC_LCA_SFT 4 +#define RT5682S_TDM_ADC_DL_MASK (0x3 << 0) #define RT5682S_TDM_ADC_DL_SFT 0 /* TDM control 2 (0x007a) */ -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 06/30] ASoC: mt6660: Keep the pm_runtime enables before component stuff in mt6660_i2c_probe
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit c4ab29b0f3a6f1e167c5a627f7cd036c1d2b7d65 ] It would be better to keep the pm_runtime enables before the IRQ and component stuff. Both of those could start triggering PM runtime events. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Link: https://lore.kernel.org/r/20221008140522.134912-1-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/mt6660.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 45e0df13afb9..b8369eeccc30 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -503,14 +503,14 @@ static int mt6660_i2c_probe(struct i2c_client *client) dev_err(chip->dev, "read chip revision fail\n"); goto probe_fail; } + pm_runtime_set_active(chip->dev); + pm_runtime_enable(chip->dev); ret = devm_snd_soc_register_component(chip->dev, &mt6660_component_driver, &mt6660_codec_dai, 1); - if (!ret) { - pm_runtime_set_active(chip->dev); - pm_runtime_enable(chip->dev); - } + if (ret) + pm_runtime_disable(chip->dev); return ret; -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 05/30] ASoC: wm8997: Revert "ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit 68ce83e3bb26feba0fcdd59667fde942b3a600a1 ] This reverts commit 41a736ac20602f64773e80f0f5b32cde1830a44a. The pm_runtime_disable is redundant when error returns in wm8997_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-4-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8997.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 77136a521605..210ad662fc26 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1161,6 +1161,9 @@ static int wm8997_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], WM8997_DIG_VU, WM8997_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + arizona_init_common(arizona); ret = arizona_init_vol_limit(arizona); @@ -1179,9 +1182,6 @@ static int wm8997_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 04/30] ASoC: wm5110: Revert "ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit 7d4e966f4cd73ff69bf06934e8e14a33fb7ef447 ] This reverts commit 86b46bf1feb83898d89a2b4a8d08d21e9ea277a7. The pm_runtime_disable is redundant when error returns in wm5110_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-3-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5110.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc634c995834..f3f4a10bf0f7 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2457,6 +2457,9 @@ static int wm5110_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5110_digital_vu[i], WM5110_DIG_VU, WM5110_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5110_adsp2_irq, wm5110); @@ -2489,9 +2492,6 @@ static int wm5110_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 03/30] ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe"
by Sasha Levin 06 Nov '22

06 Nov '22

From: Zhang Qilong <zhangqilong3(a)huawei.com> [ Upstream commit de71d7567e358effd06dfc3e2a154b25f1331c10 ] This reverts commit fcbb60820cd3008bb44334a0395e5e57ccb77329. The pm_runtime_disable is redundant when error returns in wm5102_probe, we just revert the old patch to fix it. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20221010114852.88127-2-zhangqilong3@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c09c9ac51b3e..af7d324e3352 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2099,6 +2099,9 @@ static int wm5102_probe(struct platform_device *pdev) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], WM5102_DIG_VU, WM5102_DIG_VU); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, "ADSP2 Compressed IRQ", wm5102_adsp2_irq, wm5102); @@ -2131,9 +2134,6 @@ static int wm5102_probe(struct platform_device *pdev) goto err_spk_irqs; } - pm_runtime_enable(&pdev->dev); - pm_runtime_idle(&pdev->dev); - return ret; err_spk_irqs: -- 2.35.1

1 0

[Digidesign Mbox 3] Does ucm SplitPCM add latency?
by GitHub issues - opened 06 Nov '22

06 Nov '22

alsa-project/alsa-ucm-conf issue #238 was opened from ManuLinares: https://github.com/alsa-project/alsa-ucm-conf/pull/237 I've correctly configured ucm profile, but I notice a small latency added when using UCM **vs** not using it (having the default alsa analog-surround4.0 profile) Is this the expected behavior of SplitPCM in UCM? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/238 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: add support for "Digidesign Mbox 3"
by GitHub pull_request - opened 06 Nov '22

06 Nov '22

alsa-project/alsa-ucm-conf pull request #237 was opened from ManuLinares: Add ucm2 support for "Digidesign Mbox 3", without it, alsa detects as analog-surround4.0, instead of 2 line and 2 spdif in/out. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/9944813/… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/237 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/237.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4a 00/38] timers: Use timer_shutdown*() before freeing timers
by Steven Rostedt 05 Nov '22

05 Nov '22

Back in April, I posted an RFC patch set to help mitigate a common issue where a timer gets armed just before it is freed, and when the timer goes off, it crashes in the timer code without any evidence of who the culprit was. I got side tracked and never finished up on that patch set. Since this type of crash is still our #1 crash we are seeing in the field, it has become a priority again to finish it. The last version of that patch set is here: https://lore.kernel.org/all/20221104054053.431922658@goodmis.org/ I'm calling this version 4a as it only has obvious changes were the timer that is being shutdown is in the same function where it will be freed or released, as this series should be "safe" for adding. I'll be calling the other patches 4b for the next merge window. Patch 1 fixes an issue with sunrpc/xprt where it incorrectly uses del_singleshot_timer_sync() for something that is not a oneshot timer. As this will be converted to shutdown, this needs to be fixed first. Patches 2-4 changes existing timer_shutdown() functions used locally in ARM and some drivers to better namespace names. Patch 5 implements the new timer_shutdown() and timer_shutdown_sync() functions that disable re-arming the timer after they are called. Patches 6-28 change all the locations where there's a kfree(), kfree_rcu(), kmem_cache_free() and one call_rcu() call where the RCU function frees the timer (the workqueue patch) in the same function as the del_timer{,_sync}() is called on that timer, and there's no extra exit path between the del_timer and freeing of the timer. Patches 29-32 add timer_shutdown*() on on-stack timers that are about to be released at the end of the function. Patches 33-37 add timer_shutdown*() on module timers in the module exit code. Patch 38 simply converts an open coded "shutdown" code into timer_shutdown(), as a way timer_shutdown() disables the timer is by setting that timer function to NULL. Linus, I sorted the patches this way to let you see which you would think is safe to go into this -rc. I honestly believe that they are all safe, but that's just my own opinion. This series is here: git://git.kernel.org/pub/scm/linux/kernel/git/rostedt/linux-trace.git timers-start Head SHA1: f58b516a65bac76f1bfa00126856d6c6c3d24a40 Steven Rostedt (Google) (38): SUNRPC/xprt: Use del_timer_sync() instead of del_singleshot_timer_sync() ARM: spear: Do not use timer namespace for timer_shutdown() function clocksource/drivers/arm_arch_timer: Do not use timer namespace for timer_shutdown() function clocksource/drivers/sp804: Do not use timer namespace for timer_shutdown() function timers: Add timer_shutdown_sync() and timer_shutdown() to be called before freeing timers timers: sh: Use timer_shutdown_sync() before freeing timer timers: block: Use timer_shutdown_sync() before freeing timer timers: ACPI: Use timer_shutdown_sync() before freeing timer timers: atm: Use timer_shutdown_sync() before freeing timer timers: Bluetooth: Use timer_shutdown_sync() before freeing timer timers: drm: Use timer_shutdown_sync() before freeing timer timers: HID: Use timer_shutdown_sync() before freeing timer timers: Input: Use timer_shutdown_sync() before freeing timer timers: mISDN: Use timer_shutdown_sync() before freeing timer timers: leds: Use timer_shutdown_sync() before freeing timer timers: media: Use timer_shutdown_sync() before freeing timer timers: net: Use timer_shutdown_sync() before freeing timer timers: usb: Use timer_shutdown_sync() before freeing timer timers: nfc: pn533: Use timer_shutdown_sync() before freeing timer timers: pcmcia: Use timer_shutdown_sync() before freeing timer timers: scsi: Use timer_shutdown_sync() and timer_shutdown() before freeing timer timers: tty: Use timer_shutdown_sync() before freeing timer timers: ext4: Use timer_shutdown_sync() before freeing timer timers: fs/nilfs2: Use timer_shutdown_sync() before freeing timer timers: ALSA: Use timer_shutdown_sync() before freeing timer timers: jbd2: Use timer_shutdown() before freeing timer timers: sched/psi: Use timer_shutdown_sync() before freeing timer timers: workqueue: Use timer_shutdown_sync() before freeing timer random: use timer_shutdown_sync() for on stack timers timers: dma-buf: Use timer_shutdown_sync() for on stack timers timers: drm: Use timer_shutdown_sync() for on stack timers timers: media: Use timer_shutdown_sync() for on stack timers timers: s390/cmm: Use timer_shutdown_sync() before a module is released timers: atm: Use timer_shutdown_sync() before a module is released timers: hangcheck: Use timer_shutdown_sync() before a module is released timers: ipmi: Use timer_shutdown_sync() before a module is released timers: Input: Use timer_shutdown_sync() before a module is released timers: PM: Use timer_shutdown_sync() ---- .../RCU/Design/Requirements/Requirements.rst | 2 +- Documentation/core-api/local_ops.rst | 2 +- Documentation/kernel-hacking/locking.rst | 5 ++ arch/arm/mach-spear/time.c | 8 +-- arch/s390/mm/cmm.c | 4 +- arch/sh/drivers/push-switch.c | 2 +- block/blk-iocost.c | 2 +- block/blk-iolatency.c | 2 +- block/blk-throttle.c | 2 +- block/kyber-iosched.c | 2 +- drivers/acpi/apei/ghes.c | 2 +- drivers/atm/idt77105.c | 4 +- drivers/atm/idt77252.c | 4 +- drivers/atm/iphase.c | 2 +- drivers/base/power/wakeup.c | 7 +-- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/loop.c | 2 +- drivers/block/sunvdc.c | 2 +- drivers/bluetooth/hci_bcsp.c | 2 +- drivers/bluetooth/hci_h5.c | 4 +- drivers/bluetooth/hci_qca.c | 4 +- drivers/char/hangcheck-timer.c | 4 +- drivers/char/ipmi/ipmi_msghandler.c | 2 +- drivers/char/random.c | 2 +- drivers/clocksource/arm_arch_timer.c | 12 ++-- drivers/clocksource/timer-sp804.c | 6 +- drivers/dma-buf/st-dma-fence.c | 2 +- drivers/gpu/drm/gud/gud_pipe.c | 2 +- drivers/gpu/drm/i915/i915_sw_fence.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 +- drivers/input/keyboard/locomokbd.c | 2 +- drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/mouse/alps.c | 2 +- drivers/input/serio/hil_mlc.c | 2 +- drivers/isdn/hardware/mISDN/hfcmulti.c | 5 +- drivers/isdn/mISDN/l1oip_core.c | 4 +- drivers/isdn/mISDN/timerdev.c | 4 +- drivers/leds/trigger/ledtrig-pattern.c | 2 +- drivers/leds/trigger/ledtrig-transient.c | 2 +- drivers/media/pci/ivtv/ivtv-driver.c | 2 +- drivers/media/usb/pvrusb2/pvrusb2-hdw.c | 18 +++--- drivers/media/usb/s2255/s2255drv.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 7 +-- drivers/net/ethernet/marvell/sky2.c | 2 +- drivers/net/ethernet/sun/sunvnet.c | 2 +- drivers/net/usb/sierra_net.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-dbg-tlv.c | 2 +- drivers/net/wireless/intersil/hostap/hostap_ap.c | 2 +- drivers/net/wireless/marvell/mwifiex/main.c | 2 +- drivers/net/wireless/microchip/wilc1000/hif.c | 6 +- drivers/nfc/pn533/pn533.c | 2 +- drivers/nfc/pn533/uart.c | 2 +- drivers/pcmcia/bcm63xx_pcmcia.c | 2 +- drivers/pcmcia/electra_cf.c | 2 +- drivers/pcmcia/omap_cf.c | 2 +- drivers/pcmcia/pd6729.c | 4 +- drivers/pcmcia/yenta_socket.c | 4 +- drivers/scsi/qla2xxx/qla_edif.c | 4 +- drivers/staging/media/atomisp/i2c/atomisp-lm3554.c | 2 +- drivers/tty/n_gsm.c | 2 +- drivers/tty/sysrq.c | 2 +- drivers/usb/gadget/udc/m66592-udc.c | 2 +- drivers/usb/serial/garmin_gps.c | 2 +- drivers/usb/serial/mos7840.c | 2 +- fs/ext4/super.c | 2 +- fs/jbd2/journal.c | 2 + fs/nilfs2/segment.c | 2 +- include/linux/timer.h | 64 +++++++++++++++++++--- kernel/sched/psi.c | 1 + kernel/time/timer.c | 64 ++++++++++++---------- kernel/workqueue.c | 4 +- net/802/garp.c | 2 +- net/802/mrp.c | 2 +- net/bridge/br_multicast.c | 6 +- net/bridge/br_multicast_eht.c | 4 +- net/core/gen_estimator.c | 2 +- net/core/neighbour.c | 2 + net/ipv4/inet_timewait_sock.c | 1 + net/ipv4/ipmr.c | 2 +- net/ipv6/ip6mr.c | 2 +- net/mac80211/mesh_pathtbl.c | 2 +- net/netfilter/ipset/ip_set_list_set.c | 2 +- net/netfilter/ipvs/ip_vs_lblc.c | 2 +- net/netfilter/ipvs/ip_vs_lblcr.c | 2 +- net/netfilter/xt_LED.c | 2 +- net/rxrpc/conn_object.c | 2 +- net/sched/cls_flow.c | 2 +- net/sunrpc/svc.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/discover.c | 2 +- net/tipc/monitor.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 93 files changed, 227 insertions(+), 169 deletions(-)

4 15

[PATCH] Fix kbl_rt5663_rt5514_max98927 regression
by Jason Montleon 05 Nov '22

05 Nov '22

Starting with 6.0-rc1 these messages are logged and the sound card is unavailable. Adding legacy_dai_naming to the rt5514-spi causes it to function properly again. [ 16.928454] kbl_r5514_5663_max kbl_r5514_5663_max: ASoC: CPU DAI spi-PRP0001:00 not registered [ 16.928561] platform kbl_r5514_5663_max: deferred probe pending --- sound/soc/codecs/rt5514-spi.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..362663abcb89 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -298,13 +298,14 @@ static int rt5514_spi_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver rt5514_spi_component = { - .name = DRV_NAME, - .probe = rt5514_spi_pcm_probe, - .open = rt5514_spi_pcm_open, - .hw_params = rt5514_spi_hw_params, - .hw_free = rt5514_spi_hw_free, - .pointer = rt5514_spi_pcm_pointer, - .pcm_construct = rt5514_spi_pcm_new, + .name = DRV_NAME, + .probe = rt5514_spi_pcm_probe, + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .pointer = rt5514_spi_pcm_pointer, + .pcm_construct = rt5514_spi_pcm_new, + .legacy_dai_naming = 1, }; /** -- 2.37.3

7 13

[PATCH] Fix tas2562 volume table values
by Juan Alfonso Reyes Ajenjo 05 Nov '22

05 Nov '22

This patch fixes -90dB value (0x84a3 instead of 0x695b) This patch adds missing -20dB value (0x06666666) float_vol_db_lookup must have 56 values as volume is from 0 to 55 (0-110dB) Co-developed-by: Gustavo Plaza Roma <gplazar03(a)gmail.com> Signed-off-by: Gustavo Plaza Roma <gplazar03(a)gmail.com> Signed-off-by: Juan Alfonso Reyes Ajenjo <juanalfonso.reyes(a)gmail.com> --- sound/soc/codecs/tas2562.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b486d0b..acbdf77 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -33,15 +33,16 @@ static const unsigned int float_vol_db_lookup[] = { 0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, 0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, -0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x000084a3, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, 0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, 0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, 0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, 0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, 0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, 0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, -0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, -0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +0x06666666, 0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, +0x143d1362, 0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, +0x40000000 }; struct tas2562_data { -- 2.7.4

1 0

[PATCH] patch_ca0132: add quirk for EVGA Z390 DARK
by Xian Wang 05 Nov '22

05 Nov '22

The Z390 DARK mainboard uses a CA0132 audio controller. The quirk is needed to enable surround sound and 3.5mm headphone jack handling in the front audio connector as well as in the rear of the board when in stereo mode. Page 97 of the linked manual contains instructions to setup the controller. Link: https://www.evga.com/support/manuals/files/131-CS-E399.pdf Cc: stable(a)vger.kernel.org Signed-off-by: Xian Wang <dev(a)xianwang.io> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 9580fe00cbd9..0a292bf271f2 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1306,6 +1306,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI), SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", QUIRK_R3DI), SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), -- 2.38.1

2 1

[PATCH v6 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 05 Nov '22

05 Nov '22

v6: - dmic codec driver: - Modify "FrontMic" to "Front Mic" and "RearMic" to "Rear Mic" to consisit with commit 3cfbf07c6d27 mentioned in description. v5: - Documentation: - Add #include <dt-bindings/gpio/gpio.h> in order to use GPIO_ACTIVE_HIGH. - Remove the change-id in message. v4: - Documentation: - Fix the FATAL ERROR: Unable to parse input tree. v3: - Documentation: - Add an explain example in description. - Add the pinctrl-name and pinctrl id in its example. v2: - dmic codec driver: - Remove the unnecessary use of_property_read_bool() v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 14 ++- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 102 +++++++++++++++++- 2 files changed, 114 insertions(+), 2 deletions(-) -- 2.25.1

4 7

[snd-dice] latency 13 times higher compared to FFADO (Saffire 14 PRO) with jack2
by GitHub issues - opened 05 Nov '22

05 Nov '22

alsa-project/alsa-lib issue #280 was opened from grinness: Hi, I hope this is the right place to report this (please let me know if it is not) On a 5900x, 32 GB RAM, Archlinux (kernel 6.0.2), I have been investigating crazy high latency using a pipewire-jack, see: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2790 I ended up testing jack_iodelay with jack2, comparing snd-dice and FFADO Running qjackctl jack2, ALSA backed (snd-dice) 48000Hz, 1024 samples, 2 periods: ``` > jack_iodelay new capture latency: [0, 0] new playback latency: [0, 0] Signal below threshold... Signal below threshold... new capture latency: [1024, 1024] Signal below threshold... Signal below threshold... new playback latency: [2048, 2048] 5198.818 frames 108.309 ms total roundtrip latency extra loopback latency: 2126 frames use 1063 for the backend arguments -I and -O Inv 5198.818 frames 108.309 ms total roundtrip latency extra loopback latency: 2126 frames use 1063 for the backend arguments -I and -O ?? Inv ``` Running qjackctl jack2, firewire backend (FFADO) 4800, 512 samples, 2 periods: ``` > jack_iodelay new capture latency: [0, 0] new playback latency: [0, 0] Signal below threshold... new capture latency: [1024, 1024] Signal below threshold... new playback latency: [3072, 3072] 4258.811 frames 88.725 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv 4258.812 frames 88.725 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv ``` Note that qjackctl seems to not set the number of periods correctly with firewire backend, the above playback latency suggests 3 periods instead of 2. It seems that Cadence fixes the issue, but the extra loopback latency stays at 162 frames with FFADO: ``` > jack_iodelay new capture latency: [0, 0] new playback latency: [0, 0] Signal below threshold... Signal below threshold... Signal below threshold... Signal below threshold... Signal below threshold... Signal below threshold... new capture latency: [512, 512] 8354.799 frames 174.058 ms total roundtrip latency extra loopback latency: 7842 frames use 3921 for the backend arguments -I and -O ?? Inv 8354.779 frames 174.058 ms total roundtrip latency extra loopback latency: 7842 frames use 3921 for the backend arguments -I and -O Inv Signal below threshold... Signal below threshold... new playback latency: [1024, 1024] Signal below threshold... 1698.777 frames 35.391 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv 1698.777 frames 35.391 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv 1698.777 frames 35.391 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv 1698.777 frames 35.391 ms total roundtrip latency extra loopback latency: 162 frames use 81 for the backend arguments -I and -O Inv 1698.777 frames 35.391 ms total roundtrip latency extra loopback latency: 162 frames ``` Please note that the pipewire developer pointed out that: "My problem is that the period size in the driver is set to 1024. For batch devices and a quantum of 256 this should be set to 128 to get decent latency. So, either the driver does not report itself as batch, or something else started the nodes with a higher quantum. I'm betting it's the batch thing." I also reported this in the following: https://github.com/takaswie/snd-firewire-improve/issues/47 thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/280 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFC][PATCH v3 00/33] timers: Use timer_shutdown*() before freeing timers
by Steven Rostedt 05 Nov '22

05 Nov '22

Back in April, I posted an RFC patch set to help mitigate a common issue where a timer gets armed just before it is freed, and when the timer goes off, it crashes in the timer code without any evidence of who the culprit was. I got side tracked and never finished up on that patch set. Since this type of crash is still our #1 crash we are seeing in the field, it has become a priority again to finish it. This is v3 of that patch set. Thomas Gleixner posted an untested version that makes timer->function NULL as the flag that it is shutdown. I took that code, tested it (fixed it up), added more comments, and changed the name to timer_shutdown_sync(). I also converted it to use WARN_ON_ONCE() instead of just WARN_ON() as Linus asked for. I then created a trivial coccinelle script to find where del_timer*() is called before being freed, and converted them all to timer_shutdown*() (There was a couple that still used del_timer() instead of del_timer_sync()). I also updated DEBUG_OBJECTS_TIMERS to check from where the timer is ever armed, to calling of timer_shutdown_sync(), and it will trigger if a timer is freed in between. The current way is to only check if the timer is armed, but that means it only triggers if the race condition is hit, and with experience, it's not run on enough machines to catch all of them. By triggering it from the time the timer is armed to the time it is shutdown, it catches all potential cases even if the race condition is not hit. I went though the result of the cocinelle script, and updated the locations. Some locations were caught by DEBUG_OBJECTS_TIMERS as the coccinelle script only checked for timers being freed in the same function as the del_timer*(). Ideally, I would have the first patch go into this rc cycle, which is mostly non functional as it will allow the other patches to come in via the respective subsystems in the next merge window. Changes since v2: https://lore.kernel.org/all/20221027150525.753064657@goodmis.org/ - Talking with Thomas Gleixner, he wanted a better name space and to remove the "del_" portion of the API. - Since there's now a shutdown interface that does not synchronize, to keep it closer to del_timer() and del_timer_sync(), the API is now: timer_shutdown() - same as del_timer() but deactivates the timer. timer_shutdown_sync() - same as del_timer_sync() but deactivates the timer. - Added a few more locations that got converted. git://git.kernel.org/pub/scm/linux/kernel/git/rostedt/linux-trace.git trace/timers Head SHA1: 25106f0bb7968b3e8c746a7853f44b51840746c3 Steven Rostedt (Google) (33): timers: Add timer_shutdown_sync() and timer_shutdown() to be called before freeing timers timers: s390/cmm: Use timer_shutdown_sync() before freeing timer timers: sh: Use timer_shutdown_sync() before freeing timer timers: block: Use timer_shutdown_sync() before freeing timer timers: ACPI: Use timer_shutdown_sync() before freeing timer timers: atm: Use timer_shutdown_sync() before freeing timer timers: PM: Use timer_shutdown_sync() timers: Bluetooth: Use timer_shutdown_sync() before freeing timer timers: hangcheck: Use timer_shutdown_sync() before freeing timer timers: ipmi: Use timer_shutdown_sync() before freeing timer random: use timer_shutdown_sync() before freeing timer timers: dma-buf: Use timer_shutdown_sync() before freeing timer timers: drm: Use timer_shutdown_sync() before freeing timer timers: HID: Use timer_shutdown_sync() before freeing timer timers: Input: Use timer_shutdown_sync() before freeing timer timers: mISDN: Use timer_shutdown_sync() before freeing timer timers: leds: Use timer_shutdown_sync() before freeing timer timers: media: Use timer_shutdown_sync() before freeing timer timers: net: Use timer_shutdown_sync() before freeing timer timers: usb: Use timer_shutdown_sync() before freeing timer timers: cgroup: Use timer_shutdown_sync() before freeing timer timers: workqueue: Use timer_shutdown_sync() before freeing timer timers: nfc: pn533: Use timer_shutdown_sync() before freeing timer timers: pcmcia: Use timer_shutdown_sync() before freeing timer timers: scsi: Use timer_shutdown_sync() and timer_shutdown() before freeing timer timers: tty: Use timer_shutdown_sync() before freeing timer timers: ext4: Use timer_shutdown_sync() before freeing timer timers: fs/nilfs2: Use timer_shutdown_sync() before freeing timer timers: ALSA: Use timer_shutdown_sync() before freeing timer timers: jbd2: Use timer_shutdown() before freeing timer timers: sched/psi: Use timer_shutdown_sync() before freeing timer timers: x86/mce: Use __init_timer() for resetting timers timers: Expand DEBUG_OBJECTS_TIMER to check if it ever was used ---- .../RCU/Design/Requirements/Requirements.rst | 2 +- Documentation/core-api/local_ops.rst | 2 +- Documentation/kernel-hacking/locking.rst | 5 + arch/s390/mm/cmm.c | 4 +- arch/sh/drivers/push-switch.c | 2 +- arch/x86/kernel/cpu/mce/core.c | 14 ++- block/blk-iocost.c | 2 +- block/blk-iolatency.c | 2 +- block/blk-stat.c | 2 +- block/blk-throttle.c | 2 +- block/kyber-iosched.c | 2 +- drivers/acpi/apei/ghes.c | 2 +- drivers/atm/idt77105.c | 4 +- drivers/atm/idt77252.c | 4 +- drivers/atm/iphase.c | 2 +- drivers/base/power/wakeup.c | 7 +- drivers/block/drbd/drbd_main.c | 2 +- drivers/block/loop.c | 2 +- drivers/block/sunvdc.c | 2 +- drivers/bluetooth/hci_bcsp.c | 2 +- drivers/bluetooth/hci_h5.c | 2 +- drivers/bluetooth/hci_qca.c | 4 +- drivers/char/hangcheck-timer.c | 4 +- drivers/char/ipmi/ipmi_msghandler.c | 2 +- drivers/char/ipmi/ipmi_ssif.c | 4 +- drivers/char/random.c | 2 +- drivers/dma-buf/st-dma-fence.c | 2 +- drivers/gpu/drm/gud/gud_pipe.c | 2 +- drivers/gpu/drm/i915/i915_sw_fence.c | 2 +- drivers/hid/hid-wiimote-core.c | 2 +- drivers/input/keyboard/locomokbd.c | 2 +- drivers/input/keyboard/omap-keypad.c | 2 +- drivers/input/mouse/alps.c | 2 +- drivers/input/serio/hil_mlc.c | 2 +- drivers/input/serio/hp_sdc.c | 2 +- drivers/isdn/hardware/mISDN/hfcmulti.c | 6 +- drivers/isdn/mISDN/l1oip_core.c | 4 +- drivers/isdn/mISDN/timerdev.c | 4 +- drivers/leds/trigger/ledtrig-activity.c | 2 +- drivers/leds/trigger/ledtrig-heartbeat.c | 2 +- drivers/leds/trigger/ledtrig-pattern.c | 2 +- drivers/leds/trigger/ledtrig-transient.c | 2 +- drivers/media/pci/ivtv/ivtv-driver.c | 2 +- drivers/media/usb/pvrusb2/pvrusb2-hdw.c | 18 ++-- drivers/media/usb/s2255/s2255drv.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 6 +- drivers/net/ethernet/marvell/sky2.c | 2 +- drivers/net/ethernet/sun/sunvnet.c | 2 +- drivers/net/usb/sierra_net.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-dbg-tlv.c | 2 +- drivers/net/wireless/intersil/hostap/hostap_ap.c | 2 +- drivers/net/wireless/marvell/mwifiex/main.c | 2 +- drivers/net/wireless/microchip/wilc1000/hif.c | 8 +- drivers/nfc/pn533/pn533.c | 2 +- drivers/nfc/pn533/uart.c | 2 +- drivers/pcmcia/bcm63xx_pcmcia.c | 2 +- drivers/pcmcia/electra_cf.c | 2 +- drivers/pcmcia/omap_cf.c | 2 +- drivers/pcmcia/pd6729.c | 4 +- drivers/pcmcia/yenta_socket.c | 4 +- drivers/scsi/qla2xxx/qla_edif.c | 4 +- drivers/scsi/scsi_lib.c | 1 + drivers/staging/media/atomisp/i2c/atomisp-lm3554.c | 2 +- drivers/tty/n_gsm.c | 2 +- drivers/tty/sysrq.c | 2 +- drivers/usb/gadget/udc/m66592-udc.c | 2 +- drivers/usb/serial/garmin_gps.c | 2 +- drivers/usb/serial/mos7840.c | 2 +- fs/ext4/super.c | 2 +- fs/jbd2/journal.c | 2 + fs/nilfs2/segment.c | 2 +- include/linux/timer.h | 100 +++++++++++++++++-- include/linux/workqueue.h | 4 +- kernel/cgroup/cgroup.c | 2 +- kernel/sched/psi.c | 1 + kernel/time/timer.c | 106 ++++++++++++++------- kernel/workqueue.c | 4 +- net/802/garp.c | 2 +- net/802/mrp.c | 2 +- net/bridge/br_multicast.c | 6 +- net/bridge/br_multicast_eht.c | 4 +- net/core/gen_estimator.c | 2 +- net/core/neighbour.c | 2 + net/ipv4/inet_connection_sock.c | 2 +- net/ipv4/inet_timewait_sock.c | 3 +- net/ipv4/ipmr.c | 2 +- net/ipv6/ip6mr.c | 2 +- net/mac80211/mesh_pathtbl.c | 2 +- net/netfilter/ipset/ip_set_list_set.c | 2 +- net/netfilter/ipvs/ip_vs_lblc.c | 2 +- net/netfilter/ipvs/ip_vs_lblcr.c | 2 +- net/netfilter/xt_LED.c | 2 +- net/rxrpc/conn_object.c | 2 +- net/sched/cls_flow.c | 2 +- net/sunrpc/svc.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/discover.c | 2 +- net/tipc/monitor.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 100 files changed, 310 insertions(+), 175 deletions(-)

4 10

Feedback with embedded USB audio gadget
by David Jaffe 05 Nov '22

05 Nov '22

I see that the USB Gadget has an explicit feedback node. It seems to deliver a constant (00 00 30 00) for a sample rate of 48kHz. I see that the value is initialized to a combination of the sample rate and the "pitch", which appears to be a control of the driver. Is the pitch plumbed up to the user-level through alsa somehow (if so how)? If not, is there some ALSA layer that servos the pitch to keep the host clock and the embedded clock synchronized?

1 1

Missing quirk for JBL Quantum 810 (?)
by GitHub issues - opened 04 Nov '22

04 Nov '22

alsa-project/alsa-lib issue #279 was opened from Xenorio: Hi there! Recently, I received a JBL Quantum 810 headset as a replacement for my Quantum 800 (which worked totally fine until a hardware issue). When I connected my new headset via the provided 2.4Ghz dongle, all I got was microphone input, but no sound output. Taking a look at pavucontrol (I use PipeWire on Fedora), it only has 2 configuration options (other than "off"), which are "Pro Audio" and "Mono Input". With the old one I had a lot more, including "Analog Stereo Duplex", which is what I always used. With "Pro Audio" selected, it registers one input device and 2 output devices (from what I could gather, there are 2 because of the hardware slider used for mixing game and voice chat audio). No matter which of the output devices I use, no sound is coming out, and they don't even have a volume slider. After some research, my suspicion is that all that needs to be done is assigning an already existing quirk for JBL headsets like what has been done [here](https://lore.kernel.org/alsa-devel/s5h4kb4m3ht.wl-tiwai@suse.de/T/). Although I'm in no way an expert in kernel or driver related matters, so take this with a grain of salt. I have tried tinkering with aliases in ``/etc/modprobe.d/alsa-base.conf`` but I seem to be too inexperienced to get it to do anything at all. I am able to connect the headset via bluetooth, which works fine, but this results in noticeable quality loss and missing features (for example, the hardware volume slider can't be used on bluetooth connections). For now, I'm connecting it via bluetooth **and** the dongle, setting the bluetooth connection to output only via blueman, then assigning input streams to the mic from the dongle connection and output streams to the bluetooth connection (this way, I'm at least getting full audio quality on the mic). This is definitely not a viable long-term solution though. And now, all the data: **Relevant output from lsusb** ``` Bus 001 Device 007: ID 0ecb:2069 Harman International Inc JBL Quantum810 Wireless ``` **uname -r** ``` 6.0.5-200.fc36.x86_64 ``` **alsa-info.sh** (headset only connected via dongle here) http://alsa-project.org/db/?f=db549be4f37c3c8617f567b1aaf26fd77c25e030 P.S.: I have done a full system update, tons of reboots, and the firmware on the headset is on the most recent version (installed Windows on a separate machine to get into the Quantum Engine software). In my desperation, I have even installed "alsa-*" via dnf. Issue URL : https://github.com/alsa-project/alsa-lib/issues/279 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: amd: fix ACP version typo mistake
by Syed Saba Kareem 04 Nov '22

04 Nov '22

From: syed saba kareem <syed.sabakareem(a)amd.com> Pink Sardine is based on ACP6.3 architecture. This patch fixes the typo mistake acp6.2 -> acp6.3 Signed-off-by: syed saba kareem <syed.sabakareem(a)amd.com> --- ...offset_byte.h => acp63_chip_offset_byte.h} | 2 +- sound/soc/amd/Kconfig | 4 +- sound/soc/amd/ps/{acp62.h => acp63.h} | 10 +- sound/soc/amd/ps/pci-ps.c | 132 ++++++------ sound/soc/amd/ps/ps-mach.c | 30 +-- sound/soc/amd/ps/ps-pdm-dma.c | 188 +++++++++--------- 6 files changed, 183 insertions(+), 183 deletions(-) rename include/sound/{acp62_chip_offset_byte.h => acp63_chip_offset_byte.h} (99%) rename sound/soc/amd/ps/{acp62.h => acp63.h} (90%) diff --git a/include/sound/acp62_chip_offset_byte.h b/include/sound/acp63_chip_offset_byte.h similarity index 99% rename from include/sound/acp62_chip_offset_byte.h rename to include/sound/acp63_chip_offset_byte.h index ca38f8a0966e..c9260e1640ae 100644 --- a/include/sound/acp62_chip_offset_byte.h +++ b/include/sound/acp63_chip_offset_byte.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0+ */ /* - * AMD ACP 6.2 Register Documentation + * AMD ACP 6.3 Register Documentation * * Copyright 2022 Advanced Micro Devices, Inc. */ diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 150786279257..c88ebd84bdd5 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -129,10 +129,10 @@ config SND_SOC_AMD_RPL_ACP6x If unsure select "N". config SND_SOC_AMD_PS - tristate "AMD Audio Coprocessor-v6.2 Pink Sardine support" + tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" depends on X86 && PCI && ACPI help - This option enables Audio Coprocessor i.e ACP v6.2 support on + This option enables Audio Coprocessor i.e ACP v6.3 support on AMD Pink sardine platform. By enabling this flag build will be triggered for ACP PCI driver, ACP PDM DMA driver. Say m if you have such a device. diff --git a/sound/soc/amd/ps/acp62.h b/sound/soc/amd/ps/acp63.h similarity index 90% rename from sound/soc/amd/ps/acp62.h rename to sound/soc/amd/ps/acp63.h index 8b30aefa4cd0..85f869c2229f 100644 --- a/sound/soc/amd/ps/acp62.h +++ b/sound/soc/amd/ps/acp63.h @@ -5,7 +5,7 @@ * Copyright (C) 2022 Advanced Micro Devices, Inc. All rights reserved. */ -#include <sound/acp62_chip_offset_byte.h> +#include <sound/acp63_chip_offset_byte.h> #define ACP_DEVICE_ID 0x15E2 #define ACP6x_REG_START 0x1240000 @@ -78,21 +78,21 @@ struct pdm_stream_instance { u16 channels; dma_addr_t dma_addr; u64 bytescount; - void __iomem *acp62_base; + void __iomem *acp63_base; }; struct pdm_dev_data { u32 pdm_irq; - void __iomem *acp62_base; + void __iomem *acp63_base; struct snd_pcm_substream *capture_stream; }; -static inline u32 acp62_readl(void __iomem *base_addr) +static inline u32 acp63_readl(void __iomem *base_addr) { return readl(base_addr); } -static inline void acp62_writel(u32 val, void __iomem *base_addr) +static inline void acp63_writel(u32 val, void __iomem *base_addr) { writel(val, base_addr); } diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index dff2e2376bbf..08c4b9bef055 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -15,30 +15,30 @@ #include <sound/pcm_params.h> #include <linux/pm_runtime.h> -#include "acp62.h" +#include "acp63.h" -struct acp62_dev_data { - void __iomem *acp62_base; +struct acp63_dev_data { + void __iomem *acp63_base; struct resource *res; - bool acp62_audio_mode; + bool acp63_audio_mode; struct platform_device *pdev[ACP6x_DEVS]; }; -static int acp62_power_on(void __iomem *acp_base) +static int acp63_power_on(void __iomem *acp_base) { u32 val; int timeout; - val = acp62_readl(acp_base + ACP_PGFSM_STATUS); + val = acp63_readl(acp_base + ACP_PGFSM_STATUS); if (!val) return val; if ((val & ACP_PGFSM_STATUS_MASK) != ACP_POWER_ON_IN_PROGRESS) - acp62_writel(ACP_PGFSM_CNTL_POWER_ON_MASK, acp_base + ACP_PGFSM_CONTROL); + acp63_writel(ACP_PGFSM_CNTL_POWER_ON_MASK, acp_base + ACP_PGFSM_CONTROL); timeout = 0; while (++timeout < 500) { - val = acp62_readl(acp_base + ACP_PGFSM_STATUS); + val = acp63_readl(acp_base + ACP_PGFSM_STATUS); if (!val) return 0; udelay(1); @@ -46,23 +46,23 @@ static int acp62_power_on(void __iomem *acp_base) return -ETIMEDOUT; } -static int acp62_reset(void __iomem *acp_base) +static int acp63_reset(void __iomem *acp_base) { u32 val; int timeout; - acp62_writel(1, acp_base + ACP_SOFT_RESET); + acp63_writel(1, acp_base + ACP_SOFT_RESET); timeout = 0; while (++timeout < 500) { - val = acp62_readl(acp_base + ACP_SOFT_RESET); + val = acp63_readl(acp_base + ACP_SOFT_RESET); if (val & ACP_SOFT_RESET_SOFTRESET_AUDDONE_MASK) break; cpu_relax(); } - acp62_writel(0, acp_base + ACP_SOFT_RESET); + acp63_writel(0, acp_base + ACP_SOFT_RESET); timeout = 0; while (++timeout < 500) { - val = acp62_readl(acp_base + ACP_SOFT_RESET); + val = acp63_readl(acp_base + ACP_SOFT_RESET); if (!val) return 0; cpu_relax(); @@ -70,57 +70,57 @@ static int acp62_reset(void __iomem *acp_base) return -ETIMEDOUT; } -static void acp62_enable_interrupts(void __iomem *acp_base) +static void acp63_enable_interrupts(void __iomem *acp_base) { - acp62_writel(1, acp_base + ACP_EXTERNAL_INTR_ENB); + acp63_writel(1, acp_base + ACP_EXTERNAL_INTR_ENB); } -static void acp62_disable_interrupts(void __iomem *acp_base) +static void acp63_disable_interrupts(void __iomem *acp_base) { - acp62_writel(ACP_EXT_INTR_STAT_CLEAR_MASK, acp_base + + acp63_writel(ACP_EXT_INTR_STAT_CLEAR_MASK, acp_base + ACP_EXTERNAL_INTR_STAT); - acp62_writel(0, acp_base + ACP_EXTERNAL_INTR_CNTL); - acp62_writel(0, acp_base + ACP_EXTERNAL_INTR_ENB); + acp63_writel(0, acp_base + ACP_EXTERNAL_INTR_CNTL); + acp63_writel(0, acp_base + ACP_EXTERNAL_INTR_ENB); } -static int acp62_init(void __iomem *acp_base, struct device *dev) +static int acp63_init(void __iomem *acp_base, struct device *dev) { int ret; - ret = acp62_power_on(acp_base); + ret = acp63_power_on(acp_base); if (ret) { dev_err(dev, "ACP power on failed\n"); return ret; } - acp62_writel(0x01, acp_base + ACP_CONTROL); - ret = acp62_reset(acp_base); + acp63_writel(0x01, acp_base + ACP_CONTROL); + ret = acp63_reset(acp_base); if (ret) { dev_err(dev, "ACP reset failed\n"); return ret; } - acp62_writel(0x03, acp_base + ACP_CLKMUX_SEL); - acp62_enable_interrupts(acp_base); + acp63_writel(0x03, acp_base + ACP_CLKMUX_SEL); + acp63_enable_interrupts(acp_base); return 0; } -static int acp62_deinit(void __iomem *acp_base, struct device *dev) +static int acp63_deinit(void __iomem *acp_base, struct device *dev) { int ret; - acp62_disable_interrupts(acp_base); - ret = acp62_reset(acp_base); + acp63_disable_interrupts(acp_base); + ret = acp63_reset(acp_base); if (ret) { dev_err(dev, "ACP reset failed\n"); return ret; } - acp62_writel(0, acp_base + ACP_CLKMUX_SEL); - acp62_writel(0, acp_base + ACP_CONTROL); + acp63_writel(0, acp_base + ACP_CLKMUX_SEL); + acp63_writel(0, acp_base + ACP_CONTROL); return 0; } -static irqreturn_t acp62_irq_handler(int irq, void *dev_id) +static irqreturn_t acp63_irq_handler(int irq, void *dev_id) { - struct acp62_dev_data *adata; + struct acp63_dev_data *adata; struct pdm_dev_data *ps_pdm_data; u32 val; @@ -128,10 +128,10 @@ static irqreturn_t acp62_irq_handler(int irq, void *dev_id) if (!adata) return IRQ_NONE; - val = acp62_readl(adata->acp62_base + ACP_EXTERNAL_INTR_STAT); + val = acp63_readl(adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (val & BIT(PDM_DMA_STAT)) { ps_pdm_data = dev_get_drvdata(&adata->pdev[0]->dev); - acp62_writel(BIT(PDM_DMA_STAT), adata->acp62_base + ACP_EXTERNAL_INTR_STAT); + acp63_writel(BIT(PDM_DMA_STAT), adata->acp63_base + ACP_EXTERNAL_INTR_STAT); if (ps_pdm_data->capture_stream) snd_pcm_period_elapsed(ps_pdm_data->capture_stream); return IRQ_HANDLED; @@ -139,10 +139,10 @@ static irqreturn_t acp62_irq_handler(int irq, void *dev_id) return IRQ_NONE; } -static int snd_acp62_probe(struct pci_dev *pci, +static int snd_acp63_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - struct acp62_dev_data *adata; + struct acp63_dev_data *adata; struct platform_device_info pdevinfo[ACP6x_DEVS]; int index, ret; int val = 0x00; @@ -157,7 +157,7 @@ static int snd_acp62_probe(struct pci_dev *pci, case 0x63: break; default: - dev_dbg(&pci->dev, "acp62 pci device not found\n"); + dev_dbg(&pci->dev, "acp63 pci device not found\n"); return -ENODEV; } if (pci_enable_device(pci)) { @@ -170,7 +170,7 @@ static int snd_acp62_probe(struct pci_dev *pci, dev_err(&pci->dev, "pci_request_regions failed\n"); goto disable_pci; } - adata = devm_kzalloc(&pci->dev, sizeof(struct acp62_dev_data), + adata = devm_kzalloc(&pci->dev, sizeof(struct acp63_dev_data), GFP_KERNEL); if (!adata) { ret = -ENOMEM; @@ -178,18 +178,18 @@ static int snd_acp62_probe(struct pci_dev *pci, } addr = pci_resource_start(pci, 0); - adata->acp62_base = devm_ioremap(&pci->dev, addr, + adata->acp63_base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); - if (!adata->acp62_base) { + if (!adata->acp63_base) { ret = -ENOMEM; goto release_regions; } pci_set_master(pci); pci_set_drvdata(pci, adata); - ret = acp62_init(adata->acp62_base, &pci->dev); + ret = acp63_init(adata->acp63_base, &pci->dev); if (ret) goto release_regions; - val = acp62_readl(adata->acp62_base + ACP_PIN_CONFIG); + val = acp63_readl(adata->acp63_base + ACP_PIN_CONFIG); switch (val) { case ACP_CONFIG_0: case ACP_CONFIG_1: @@ -220,7 +220,7 @@ static int snd_acp62_probe(struct pci_dev *pci, adata->res->flags = IORESOURCE_MEM; adata->res->start = addr; adata->res->end = addr + (ACP6x_REG_END - ACP6x_REG_START); - adata->acp62_audio_mode = ACP6x_PDM_MODE; + adata->acp63_audio_mode = ACP6x_PDM_MODE; memset(&pdevinfo, 0, sizeof(pdevinfo)); pdevinfo[0].name = "acp_ps_pdm_dma"; @@ -248,7 +248,7 @@ static int snd_acp62_probe(struct pci_dev *pci, ret = PTR_ERR(adata->pdev[index]); goto unregister_devs; } - ret = devm_request_irq(&pci->dev, pci->irq, acp62_irq_handler, + ret = devm_request_irq(&pci->dev, pci->irq, acp63_irq_handler, irqflags, "ACP_PCI_IRQ", adata); if (ret) { dev_err(&pci->dev, "ACP PCI IRQ request failed\n"); @@ -267,7 +267,7 @@ static int snd_acp62_probe(struct pci_dev *pci, for (--index; index >= 0; index--) platform_device_unregister(adata->pdev[index]); de_init: - if (acp62_deinit(adata->acp62_base, &pci->dev)) + if (acp63_deinit(adata->acp63_base, &pci->dev)) dev_err(&pci->dev, "ACP de-init failed\n"); release_regions: pci_release_regions(pci); @@ -277,46 +277,46 @@ static int snd_acp62_probe(struct pci_dev *pci, return ret; } -static int __maybe_unused snd_acp62_suspend(struct device *dev) +static int __maybe_unused snd_acp63_suspend(struct device *dev) { - struct acp62_dev_data *adata; + struct acp63_dev_data *adata; int ret; adata = dev_get_drvdata(dev); - ret = acp62_deinit(adata->acp62_base, dev); + ret = acp63_deinit(adata->acp63_base, dev); if (ret) dev_err(dev, "ACP de-init failed\n"); return ret; } -static int __maybe_unused snd_acp62_resume(struct device *dev) +static int __maybe_unused snd_acp63_resume(struct device *dev) { - struct acp62_dev_data *adata; + struct acp63_dev_data *adata; int ret; adata = dev_get_drvdata(dev); - ret = acp62_init(adata->acp62_base, dev); + ret = acp63_init(adata->acp63_base, dev); if (ret) dev_err(dev, "ACP init failed\n"); return ret; } -static const struct dev_pm_ops acp62_pm_ops = { - SET_RUNTIME_PM_OPS(snd_acp62_suspend, snd_acp62_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(snd_acp62_suspend, snd_acp62_resume) +static const struct dev_pm_ops acp63_pm_ops = { + SET_RUNTIME_PM_OPS(snd_acp63_suspend, snd_acp63_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(snd_acp63_suspend, snd_acp63_resume) }; -static void snd_acp62_remove(struct pci_dev *pci) +static void snd_acp63_remove(struct pci_dev *pci) { - struct acp62_dev_data *adata; + struct acp63_dev_data *adata; int ret, index; adata = pci_get_drvdata(pci); - if (adata->acp62_audio_mode == ACP6x_PDM_MODE) { + if (adata->acp63_audio_mode == ACP6x_PDM_MODE) { for (index = 0; index < ACP6x_DEVS; index++) platform_device_unregister(adata->pdev[index]); } - ret = acp62_deinit(adata->acp62_base, &pci->dev); + ret = acp63_deinit(adata->acp63_base, &pci->dev); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); pm_runtime_forbid(&pci->dev); @@ -325,25 +325,25 @@ static void snd_acp62_remove(struct pci_dev *pci) pci_disable_device(pci); } -static const struct pci_device_id snd_acp62_ids[] = { +static const struct pci_device_id snd_acp63_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_DEVICE_ID), .class = PCI_CLASS_MULTIMEDIA_OTHER << 8, .class_mask = 0xffffff }, { 0, }, }; -MODULE_DEVICE_TABLE(pci, snd_acp62_ids); +MODULE_DEVICE_TABLE(pci, snd_acp63_ids); -static struct pci_driver ps_acp62_driver = { +static struct pci_driver ps_acp63_driver = { .name = KBUILD_MODNAME, - .id_table = snd_acp62_ids, - .probe = snd_acp62_probe, - .remove = snd_acp62_remove, + .id_table = snd_acp63_ids, + .probe = snd_acp63_probe, + .remove = snd_acp63_remove, .driver = { - .pm = &acp62_pm_ops, + .pm = &acp63_pm_ops, } }; -module_pci_driver(ps_acp62_driver); +module_pci_driver(ps_acp63_driver); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); MODULE_AUTHOR("Syed.SabaKareem(a)amd.com"); diff --git a/sound/soc/amd/ps/ps-mach.c b/sound/soc/amd/ps/ps-mach.c index b3e97093481d..3ffbe4fdafdf 100644 --- a/sound/soc/amd/ps/ps-mach.c +++ b/sound/soc/amd/ps/ps-mach.c @@ -13,11 +13,11 @@ #include <linux/io.h> #include <linux/dmi.h> -#include "acp62.h" +#include "acp63.h" #define DRV_NAME "acp_ps_mach" -SND_SOC_DAILINK_DEF(acp62_pdm, +SND_SOC_DAILINK_DEF(acp63_pdm, DAILINK_COMP_ARRAY(COMP_CPU("acp_ps_pdm_dma.0"))); SND_SOC_DAILINK_DEF(dmic_codec, @@ -27,31 +27,31 @@ SND_SOC_DAILINK_DEF(dmic_codec, SND_SOC_DAILINK_DEF(pdm_platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_ps_pdm_dma.0"))); -static struct snd_soc_dai_link acp62_dai_pdm[] = { +static struct snd_soc_dai_link acp63_dai_pdm[] = { { - .name = "acp62-dmic-capture", + .name = "acp63-dmic-capture", .stream_name = "DMIC capture", .capture_only = 1, - SND_SOC_DAILINK_REG(acp62_pdm, dmic_codec, pdm_platform), + SND_SOC_DAILINK_REG(acp63_pdm, dmic_codec, pdm_platform), }, }; -static struct snd_soc_card acp62_card = { - .name = "acp62", +static struct snd_soc_card acp63_card = { + .name = "acp63", .owner = THIS_MODULE, - .dai_link = acp62_dai_pdm, + .dai_link = acp63_dai_pdm, .num_links = 1, }; -static int acp62_probe(struct platform_device *pdev) +static int acp63_probe(struct platform_device *pdev) { - struct acp62_pdm *machine = NULL; + struct acp63_pdm *machine = NULL; struct snd_soc_card *card; int ret; - platform_set_drvdata(pdev, &acp62_card); + platform_set_drvdata(pdev, &acp63_card); card = platform_get_drvdata(pdev); - acp62_card.dev = &pdev->dev; + acp63_card.dev = &pdev->dev; snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); @@ -64,15 +64,15 @@ static int acp62_probe(struct platform_device *pdev) return 0; } -static struct platform_driver acp62_mach_driver = { +static struct platform_driver acp63_mach_driver = { .driver = { .name = "acp_ps_mach", .pm = &snd_soc_pm_ops, }, - .probe = acp62_probe, + .probe = acp63_probe, }; -module_platform_driver(acp62_mach_driver); +module_platform_driver(acp63_mach_driver); MODULE_AUTHOR("Syed.SabaKareem(a)amd.com"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index b207b726cd82..eea71a9d2ef1 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -14,11 +14,11 @@ #include <sound/soc-dai.h> #include <linux/pm_runtime.h> -#include "acp62.h" +#include "acp63.h" #define DRV_NAME "acp_ps_pdm_dma" -static const struct snd_pcm_hardware acp62_pdm_hardware_capture = { +static const struct snd_pcm_hardware acp63_pdm_hardware_capture = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -37,61 +37,61 @@ static const struct snd_pcm_hardware acp62_pdm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; -static void acp62_init_pdm_ring_buffer(u32 physical_addr, u32 buffer_size, +static void acp63_init_pdm_ring_buffer(u32 physical_addr, u32 buffer_size, u32 watermark_size, void __iomem *acp_base) { - acp62_writel(physical_addr, acp_base + ACP_WOV_RX_RINGBUFADDR); - acp62_writel(buffer_size, acp_base + ACP_WOV_RX_RINGBUFSIZE); - acp62_writel(watermark_size, acp_base + ACP_WOV_RX_INTR_WATERMARK_SIZE); - acp62_writel(0x01, acp_base + ACPAXI2AXI_ATU_CTRL); + acp63_writel(physical_addr, acp_base + ACP_WOV_RX_RINGBUFADDR); + acp63_writel(buffer_size, acp_base + ACP_WOV_RX_RINGBUFSIZE); + acp63_writel(watermark_size, acp_base + ACP_WOV_RX_INTR_WATERMARK_SIZE); + acp63_writel(0x01, acp_base + ACPAXI2AXI_ATU_CTRL); } -static void acp62_enable_pdm_clock(void __iomem *acp_base) +static void acp63_enable_pdm_clock(void __iomem *acp_base) { u32 pdm_clk_enable, pdm_ctrl; pdm_clk_enable = ACP_PDM_CLK_FREQ_MASK; pdm_ctrl = 0x00; - acp62_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL); - pdm_ctrl = acp62_readl(acp_base + ACP_WOV_MISC_CTRL); + acp63_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL); + pdm_ctrl = acp63_readl(acp_base + ACP_WOV_MISC_CTRL); pdm_ctrl |= ACP_WOV_MISC_CTRL_MASK; - acp62_writel(pdm_ctrl, acp_base + ACP_WOV_MISC_CTRL); + acp63_writel(pdm_ctrl, acp_base + ACP_WOV_MISC_CTRL); } -static void acp62_enable_pdm_interrupts(void __iomem *acp_base) +static void acp63_enable_pdm_interrupts(void __iomem *acp_base) { u32 ext_int_ctrl; - ext_int_ctrl = acp62_readl(acp_base + ACP_EXTERNAL_INTR_CNTL); + ext_int_ctrl = acp63_readl(acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl |= PDM_DMA_INTR_MASK; - acp62_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL); + acp63_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL); } -static void acp62_disable_pdm_interrupts(void __iomem *acp_base) +static void acp63_disable_pdm_interrupts(void __iomem *acp_base) { u32 ext_int_ctrl; - ext_int_ctrl = acp62_readl(acp_base + ACP_EXTERNAL_INTR_CNTL); + ext_int_ctrl = acp63_readl(acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl &= ~PDM_DMA_INTR_MASK; - acp62_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL); + acp63_writel(ext_int_ctrl, acp_base + ACP_EXTERNAL_INTR_CNTL); } -static bool acp62_check_pdm_dma_status(void __iomem *acp_base) +static bool acp63_check_pdm_dma_status(void __iomem *acp_base) { bool pdm_dma_status; u32 pdm_enable, pdm_dma_enable; pdm_dma_status = false; - pdm_enable = acp62_readl(acp_base + ACP_WOV_PDM_ENABLE); - pdm_dma_enable = acp62_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); + pdm_enable = acp63_readl(acp_base + ACP_WOV_PDM_ENABLE); + pdm_dma_enable = acp63_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if ((pdm_enable & ACP_PDM_ENABLE) && (pdm_dma_enable & ACP_PDM_DMA_EN_STATUS)) pdm_dma_status = true; return pdm_dma_status; } -static int acp62_start_pdm_dma(void __iomem *acp_base) +static int acp63_start_pdm_dma(void __iomem *acp_base) { u32 pdm_enable; u32 pdm_dma_enable; @@ -100,12 +100,12 @@ static int acp62_start_pdm_dma(void __iomem *acp_base) pdm_enable = 0x01; pdm_dma_enable = 0x01; - acp62_enable_pdm_clock(acp_base); - acp62_writel(pdm_enable, acp_base + ACP_WOV_PDM_ENABLE); - acp62_writel(pdm_dma_enable, acp_base + ACP_WOV_PDM_DMA_ENABLE); + acp63_enable_pdm_clock(acp_base); + acp63_writel(pdm_enable, acp_base + ACP_WOV_PDM_ENABLE); + acp63_writel(pdm_dma_enable, acp_base + ACP_WOV_PDM_DMA_ENABLE); timeout = 0; while (++timeout < ACP_COUNTER) { - pdm_dma_enable = acp62_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); + pdm_dma_enable = acp63_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if ((pdm_dma_enable & 0x02) == ACP_PDM_DMA_EN_STATUS) return 0; udelay(DELAY_US); @@ -113,7 +113,7 @@ static int acp62_start_pdm_dma(void __iomem *acp_base) return -ETIMEDOUT; } -static int acp62_stop_pdm_dma(void __iomem *acp_base) +static int acp63_stop_pdm_dma(void __iomem *acp_base) { u32 pdm_enable, pdm_dma_enable; int timeout; @@ -121,14 +121,14 @@ static int acp62_stop_pdm_dma(void __iomem *acp_base) pdm_enable = 0x00; pdm_dma_enable = 0x00; - pdm_enable = acp62_readl(acp_base + ACP_WOV_PDM_ENABLE); - pdm_dma_enable = acp62_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); + pdm_enable = acp63_readl(acp_base + ACP_WOV_PDM_ENABLE); + pdm_dma_enable = acp63_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if (pdm_dma_enable & 0x01) { pdm_dma_enable = 0x02; - acp62_writel(pdm_dma_enable, acp_base + ACP_WOV_PDM_DMA_ENABLE); + acp63_writel(pdm_dma_enable, acp_base + ACP_WOV_PDM_DMA_ENABLE); timeout = 0; while (++timeout < ACP_COUNTER) { - pdm_dma_enable = acp62_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); + pdm_dma_enable = acp63_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if ((pdm_dma_enable & 0x02) == 0x00) break; udelay(DELAY_US); @@ -138,13 +138,13 @@ static int acp62_stop_pdm_dma(void __iomem *acp_base) } if (pdm_enable == ACP_PDM_ENABLE) { pdm_enable = ACP_PDM_DISABLE; - acp62_writel(pdm_enable, acp_base + ACP_WOV_PDM_ENABLE); + acp63_writel(pdm_enable, acp_base + ACP_WOV_PDM_ENABLE); } - acp62_writel(0x01, acp_base + ACP_WOV_PDM_FIFO_FLUSH); + acp63_writel(0x01, acp_base + ACP_WOV_PDM_FIFO_FLUSH); return 0; } -static void acp62_config_dma(struct pdm_stream_instance *rtd, int direction) +static void acp63_config_dma(struct pdm_stream_instance *rtd, int direction) { u16 page_idx; u32 low, high, val; @@ -154,24 +154,24 @@ static void acp62_config_dma(struct pdm_stream_instance *rtd, int direction) val = PDM_PTE_OFFSET; /* Group Enable */ - acp62_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp62_base + + acp63_writel(ACP_SRAM_PTE_OFFSET | BIT(31), rtd->acp63_base + ACPAXI2AXI_ATU_BASE_ADDR_GRP_1); - acp62_writel(PAGE_SIZE_4K_ENABLE, rtd->acp62_base + + acp63_writel(PAGE_SIZE_4K_ENABLE, rtd->acp63_base + ACPAXI2AXI_ATU_PAGE_SIZE_GRP_1); for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ low = lower_32_bits(addr); high = upper_32_bits(addr); - acp62_writel(low, rtd->acp62_base + ACP_SCRATCH_REG_0 + val); + acp63_writel(low, rtd->acp63_base + ACP_SCRATCH_REG_0 + val); high |= BIT(31); - acp62_writel(high, rtd->acp62_base + ACP_SCRATCH_REG_0 + val + 4); + acp63_writel(high, rtd->acp63_base + ACP_SCRATCH_REG_0 + val + 4); val += 8; addr += PAGE_SIZE; } } -static int acp62_pdm_dma_open(struct snd_soc_component *component, +static int acp63_pdm_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; @@ -186,7 +186,7 @@ static int acp62_pdm_dma_open(struct snd_soc_component *component, return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - runtime->hw = acp62_pdm_hardware_capture; + runtime->hw = acp63_pdm_hardware_capture; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -196,17 +196,17 @@ static int acp62_pdm_dma_open(struct snd_soc_component *component, return ret; } - acp62_enable_pdm_interrupts(adata->acp62_base); + acp63_enable_pdm_interrupts(adata->acp63_base); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) adata->capture_stream = substream; - pdm_data->acp62_base = adata->acp62_base; + pdm_data->acp63_base = adata->acp63_base; runtime->private_data = pdm_data; return ret; } -static int acp62_pdm_dma_hw_params(struct snd_soc_component *component, +static int acp63_pdm_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -220,26 +220,26 @@ static int acp62_pdm_dma_hw_params(struct snd_soc_component *component, period_bytes = params_period_bytes(params); rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); - acp62_config_dma(rtd, substream->stream); - acp62_init_pdm_ring_buffer(PDM_MEM_WINDOW_START, size, - period_bytes, rtd->acp62_base); + acp63_config_dma(rtd, substream->stream); + acp63_init_pdm_ring_buffer(PDM_MEM_WINDOW_START, size, + period_bytes, rtd->acp63_base); return 0; } -static u64 acp62_pdm_get_byte_count(struct pdm_stream_instance *rtd, +static u64 acp63_pdm_get_byte_count(struct pdm_stream_instance *rtd, int direction) { u32 high, low; u64 byte_count; - high = acp62_readl(rtd->acp62_base + ACP_WOV_RX_LINEARPOSITIONCNTR_HIGH); + high = acp63_readl(rtd->acp63_base + ACP_WOV_RX_LINEARPOSITIONCNTR_HIGH); byte_count = high; - low = acp62_readl(rtd->acp62_base + ACP_WOV_RX_LINEARPOSITIONCNTR_LOW); + low = acp63_readl(rtd->acp63_base + ACP_WOV_RX_LINEARPOSITIONCNTR_LOW); byte_count = (byte_count << 32) | low; return byte_count; } -static snd_pcm_uframes_t acp62_pdm_dma_pointer(struct snd_soc_component *comp, +static snd_pcm_uframes_t acp63_pdm_dma_pointer(struct snd_soc_component *comp, struct snd_pcm_substream *stream) { struct pdm_stream_instance *rtd; @@ -249,14 +249,14 @@ static snd_pcm_uframes_t acp62_pdm_dma_pointer(struct snd_soc_component *comp, rtd = stream->runtime->private_data; buffersize = frames_to_bytes(stream->runtime, stream->runtime->buffer_size); - bytescount = acp62_pdm_get_byte_count(rtd, stream->stream); + bytescount = acp63_pdm_get_byte_count(rtd, stream->stream); if (bytescount > rtd->bytescount) bytescount -= rtd->bytescount; pos = do_div(bytescount, buffersize); return bytes_to_frames(stream->runtime, pos); } -static int acp62_pdm_dma_new(struct snd_soc_component *component, +static int acp63_pdm_dma_new(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { struct device *parent = component->dev->parent; @@ -266,19 +266,19 @@ static int acp62_pdm_dma_new(struct snd_soc_component *component, return 0; } -static int acp62_pdm_dma_close(struct snd_soc_component *component, +static int acp63_pdm_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct pdm_dev_data *adata = dev_get_drvdata(component->dev); struct snd_pcm_runtime *runtime = substream->runtime; - acp62_disable_pdm_interrupts(adata->acp62_base); + acp63_disable_pdm_interrupts(adata->acp63_base); adata->capture_stream = NULL; kfree(runtime->private_data); return 0; } -static int acp62_pdm_dai_trigger(struct snd_pcm_substream *substream, +static int acp63_pdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct pdm_stream_instance *rtd; @@ -299,20 +299,20 @@ static int acp62_pdm_dai_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - acp62_writel(ch_mask, rtd->acp62_base + ACP_WOV_PDM_NO_OF_CHANNELS); - acp62_writel(PDM_DECIMATION_FACTOR, rtd->acp62_base + + acp63_writel(ch_mask, rtd->acp63_base + ACP_WOV_PDM_NO_OF_CHANNELS); + acp63_writel(PDM_DECIMATION_FACTOR, rtd->acp63_base + ACP_WOV_PDM_DECIMATION_FACTOR); - rtd->bytescount = acp62_pdm_get_byte_count(rtd, substream->stream); - pdm_status = acp62_check_pdm_dma_status(rtd->acp62_base); + rtd->bytescount = acp63_pdm_get_byte_count(rtd, substream->stream); + pdm_status = acp63_check_pdm_dma_status(rtd->acp63_base); if (!pdm_status) - ret = acp62_start_pdm_dma(rtd->acp62_base); + ret = acp63_start_pdm_dma(rtd->acp63_base); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - pdm_status = acp62_check_pdm_dma_status(rtd->acp62_base); + pdm_status = acp63_check_pdm_dma_status(rtd->acp63_base); if (pdm_status) - ret = acp62_stop_pdm_dma(rtd->acp62_base); + ret = acp63_stop_pdm_dma(rtd->acp63_base); break; default: ret = -EINVAL; @@ -321,11 +321,11 @@ static int acp62_pdm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -static const struct snd_soc_dai_ops acp62_pdm_dai_ops = { - .trigger = acp62_pdm_dai_trigger, +static const struct snd_soc_dai_ops acp63_pdm_dai_ops = { + .trigger = acp63_pdm_dai_trigger, }; -static struct snd_soc_dai_driver acp62_pdm_dai_driver = { +static struct snd_soc_dai_driver acp63_pdm_dai_driver = { .name = "acp_ps_pdm_dma.0", .capture = { .rates = SNDRV_PCM_RATE_48000, @@ -335,19 +335,19 @@ static struct snd_soc_dai_driver acp62_pdm_dai_driver = { .rate_min = 48000, .rate_max = 48000, }, - .ops = &acp62_pdm_dai_ops, + .ops = &acp63_pdm_dai_ops, }; -static const struct snd_soc_component_driver acp62_pdm_component = { +static const struct snd_soc_component_driver acp63_pdm_component = { .name = DRV_NAME, - .open = acp62_pdm_dma_open, - .close = acp62_pdm_dma_close, - .hw_params = acp62_pdm_dma_hw_params, - .pointer = acp62_pdm_dma_pointer, - .pcm_construct = acp62_pdm_dma_new, + .open = acp63_pdm_dma_open, + .close = acp63_pdm_dma_close, + .hw_params = acp63_pdm_dma_hw_params, + .pointer = acp63_pdm_dma_pointer, + .pcm_construct = acp63_pdm_dma_new, }; -static int acp62_pdm_audio_probe(struct platform_device *pdev) +static int acp63_pdm_audio_probe(struct platform_device *pdev) { struct resource *res; struct pdm_dev_data *adata; @@ -363,16 +363,16 @@ static int acp62_pdm_audio_probe(struct platform_device *pdev) if (!adata) return -ENOMEM; - adata->acp62_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!adata->acp62_base) + adata->acp63_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); + if (!adata->acp63_base) return -ENOMEM; adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, - &acp62_pdm_component, - &acp62_pdm_dai_driver, 1); + &acp63_pdm_component, + &acp63_pdm_dai_driver, 1); if (status) { dev_err(&pdev->dev, "Fail to register acp pdm dai\n"); @@ -385,13 +385,13 @@ static int acp62_pdm_audio_probe(struct platform_device *pdev) return 0; } -static int acp62_pdm_audio_remove(struct platform_device *pdev) +static int acp63_pdm_audio_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); return 0; } -static int __maybe_unused acp62_pdm_resume(struct device *dev) +static int __maybe_unused acp63_pdm_resume(struct device *dev) { struct pdm_dev_data *adata; struct snd_pcm_runtime *runtime; @@ -404,47 +404,47 @@ static int __maybe_unused acp62_pdm_resume(struct device *dev) rtd = runtime->private_data; period_bytes = frames_to_bytes(runtime, runtime->period_size); buffer_len = frames_to_bytes(runtime, runtime->buffer_size); - acp62_config_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); - acp62_init_pdm_ring_buffer(PDM_MEM_WINDOW_START, buffer_len, - period_bytes, adata->acp62_base); + acp63_config_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + acp63_init_pdm_ring_buffer(PDM_MEM_WINDOW_START, buffer_len, + period_bytes, adata->acp63_base); } - acp62_enable_pdm_interrupts(adata->acp62_base); + acp63_enable_pdm_interrupts(adata->acp63_base); return 0; } -static int __maybe_unused acp62_pdm_suspend(struct device *dev) +static int __maybe_unused acp63_pdm_suspend(struct device *dev) { struct pdm_dev_data *adata; adata = dev_get_drvdata(dev); - acp62_disable_pdm_interrupts(adata->acp62_base); + acp63_disable_pdm_interrupts(adata->acp63_base); return 0; } -static int __maybe_unused acp62_pdm_runtime_resume(struct device *dev) +static int __maybe_unused acp63_pdm_runtime_resume(struct device *dev) { struct pdm_dev_data *adata; adata = dev_get_drvdata(dev); - acp62_enable_pdm_interrupts(adata->acp62_base); + acp63_enable_pdm_interrupts(adata->acp63_base); return 0; } -static const struct dev_pm_ops acp62_pdm_pm_ops = { - SET_RUNTIME_PM_OPS(acp62_pdm_suspend, acp62_pdm_runtime_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(acp62_pdm_suspend, acp62_pdm_resume) +static const struct dev_pm_ops acp63_pdm_pm_ops = { + SET_RUNTIME_PM_OPS(acp63_pdm_suspend, acp63_pdm_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(acp63_pdm_suspend, acp63_pdm_resume) }; -static struct platform_driver acp62_pdm_dma_driver = { - .probe = acp62_pdm_audio_probe, - .remove = acp62_pdm_audio_remove, +static struct platform_driver acp63_pdm_dma_driver = { + .probe = acp63_pdm_audio_probe, + .remove = acp63_pdm_audio_remove, .driver = { .name = "acp_ps_pdm_dma", - .pm = &acp62_pdm_pm_ops, + .pm = &acp63_pdm_pm_ops, }, }; -module_platform_driver(acp62_pdm_dma_driver); +module_platform_driver(acp63_pdm_dma_driver); MODULE_AUTHOR("Syed.SabaKareem(a)amd.com"); MODULE_DESCRIPTION("AMD PINK SARDINE PDM Driver"); -- 2.25.1

3 2

Re: regression with SG DMA buf allocations and IOMMU in low-mem
by Kai Vehmanen 04 Nov '22

04 Nov '22

Hi Takashi and list, I've been debugging a SOF DSP load fail on VT-D/IOMMU systems [1] and that brought me to these two commits from you: "ALSA: memalloc: Add fallback SG-buffer allocations for x86" and "ALSA: memalloc: Revive x86-specific WC page allocations again" We have an allocation with: snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); .. but in rare low-memory cases, the dma_alloc_noncontiguous() call will fail in core/memalloc.c and we go to the fallback path. So we go to snd_dma_sg_fallback_alloc(), but here it seems something is missing. The pages are not allocated with DMA mapping API anymore, so IOMMU won't know about the memory and in our case the DSP load will fail to a IOMMU fault. Looking at 5.15 code, the fallback looks very different, but in fallback we still use the DMA mapping API via snd_dma_dev_alloc() so even if we go to fallback path, mapping is still ok. Any ideas which way this should be fixed? Given the many changes I thought it's better to ask early on the list about this. [1] https://github.com/thesofproject/linux/issues/3915 Br, Kai

1 0

[PATCH] spi: amd: Fix SPI_SPD7 value
by Vitaly Rodionov 04 Nov '22

04 Nov '22

According to data sheet SPI_SPD7 should be set to 7. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- drivers/spi/spi-amd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/spi/spi-amd.c b/drivers/spi/spi-amd.c index e23121456c70..bfc3ab5f39ea 100644 --- a/drivers/spi/spi-amd.c +++ b/drivers/spi/spi-amd.c @@ -65,7 +65,7 @@ enum amd_spi_speed { F_16_66MHz, F_100MHz, F_800KHz, - SPI_SPD7, + SPI_SPD7 = 0x7, F_50MHz = 0x4, F_4MHz = 0x32, F_3_17MHz = 0x3F -- 2.34.1

2 1

[PATCH v4 1/2] ASoC: rt5514: fix legacy dai naming
by Jason Montleon 04 Nov '22

04 Nov '22

Starting with 6.0-rc1 these messages are logged and the sound card is unavailable. Adding legacy_dai_naming to the rt5514-spi causes it to function properly again. [ 16.928454] kbl_r5514_5663_max kbl_r5514_5663_max: ASoC: CPU DAI spi-PRP0001:00 not registered [ 16.928561] platform kbl_r5514_5663_max: deferred probe pending Fixes: fc34ece41f71 ("ASoC: Refactor non_legacy_dai_naming flag") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216641 Signed-off-by: Jason Montleon <jmontleo(a)redhat.com> --- sound/soc/codecs/rt5514-spi.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..362663abcb89 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -298,13 +298,14 @@ static int rt5514_spi_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver rt5514_spi_component = { - .name = DRV_NAME, - .probe = rt5514_spi_pcm_probe, - .open = rt5514_spi_pcm_open, - .hw_params = rt5514_spi_hw_params, - .hw_free = rt5514_spi_hw_free, - .pointer = rt5514_spi_pcm_pointer, - .pcm_construct = rt5514_spi_pcm_new, + .name = DRV_NAME, + .probe = rt5514_spi_pcm_probe, + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .pointer = rt5514_spi_pcm_pointer, + .pcm_construct = rt5514_spi_pcm_new, + .legacy_dai_naming = 1, }; /** -- 2.37.3

3 4

[PATCH] ASoC: dt-bindings: fsl-sai: Convert to YAML DT schema
by Marek Vasut 04 Nov '22

04 Nov '22

Convert the SAI bindings to YAML DT schema to permit validation. Add Shengjiu as maintainer, derived from sound/soc/fsl/fsl_sai.c get_maintainer result. Describe existing used combinations of compatible strings, add the missing imx7d-sai compatible string which is used on i.MX7 . Properties lsb-first, fsl,sai-synchronous-rx, fsl,sai-asynchronous, fsl,dataline are no longer listed as required, since those are clearly optional, even the description says so, so does their usage. Fix the undefined edma channel macro per arch/arm/boot/dts/vfxxx.dtsi , use the value itself just like in the vfxxx.dtsi . Document interrupts property, which was previously undocumented, but it is required property of this IP. Document #sound-sai-cells, which should be zero for this IP. Document fsl,imx6ul-iomuxc-gpr and its dependency on MX6UL and fsl,sai-mclk-direction-output . Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: alsa-devel(a)alsa-project.org To: devicetree(a)vger.kernel.org --- .../devicetree/bindings/sound/fsl-sai.yaml | 188 ++++++++++++++++++ 1 file changed, 188 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.yaml b/Documentation/devicetree/bindings/sound/fsl-sai.yaml new file mode 100644 index 0000000000000..e6620a127f419 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-sai.yaml @@ -0,0 +1,188 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)gmail.com> + +description: + The SAI is based on I2S module that used communicating with audio + codecs, which provides a synchronous audio interface that supports + fullduplex serial interfaces with frame synchronization such as I2S, + AC97, TDM, and codec/DSP interfaces. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx6ul-sai + - fsl,imx7d-sai + - const: fsl,imx6sx-sai + + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + - items: + - enum: + - fsl,imx6sx-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - fsl,vf610-sai + + reg: + maxItems: 1 + + clocks: + minItems: 4 + maxItems: 6 + + clock-names: + minItems: 4 + items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + + dmas: + maxItems: 2 + + dma-names: + items: + - const: tx + - const: rx + + interrupts: + maxItems: 1 + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted first for the + fifo data. If this property is absent, the MSB is transmitted first + as default, or the LSB is transmitted first. + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating that SAI will + work in the synchronous mode (sync Tx with Rx) which means both + the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are + absent, the default synchronous mode (sync Rx with Tx) will + be used, which means both transmitter and receiver will send + and receive data by following clocks of transmitter. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If present, indicating that SAI will work in the asynchronous + mode, which means both transmitter and receiver will send and + receive data by following their own bit clocks and frame sync + clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are + absent, the default synchronous mode (sync Rx with Tx) will + be used, which means both transmitter and receiver will send + and receive data by following clocks of transmitter. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint8-array + description: | + Configure the dataline. It has 3 values for each configuration: + first one means the type: I2S(1) or PDM(2) + second one is dataline mask for 'rx' + third one is dataline mask for 'tx'. + for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + means I2S type rx mask is 0xff, tx mask is 0xff, PDM type + rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + If present, indicates that SAI will output the SAI MCLK clock. + + fsl,imx6ul-iomuxc-gpr: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to MX6UL IOMUXC GPR shared register file. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Required if all the SAI registers are big-endian rather + than little-endian. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - dmas + - dma-names + - interrupts + +unevaluatedProperties: false + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx6ul-sai + then: + dependencies: + fsl,imx6ul-iomuxc-gpr: [ "fsl,sai-mclk-direction-output" ] + + - if: + not: + properties: + compatible: + contains: + enum: + - fsl,imx6ul-sai + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - fsl,imx8mq-sai + then: + properties: + fsl,sai-mclk-direction-output: false + +examples: + - | + #include <dt-bindings/clock/vf610-clock.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, <&edma0 0 20>; + big-endian; + lsb-first; + }; -- 2.35.1

3 3

[PATCH 00/30] ARM: pxa: remove all unused boards&drivers
by Arnd Bergmann 04 Nov '22

04 Nov '22

From: Arnd Bergmann <arnd(a)arndb.de> Most of the legacy PXA board files were marked as unused in linux-5.19 and can get removed in linux-6.2. There is support for pxa250/pxa270/pxa300 using devicetree already, which supports a number of boards, but progress on converting the remaining ones has stalled over the past few years. The two boards that are left in the tree for now are the three 'sharpsl' variants (spitz/akita/borzoi) and the 'gumstix' family of machines. Both of these are supported by qemu, which can be helpful for completing the DT conversion. Only PXA25x and PXA27x are left with full boardfile support, so PXA300 can get a lot simpler while still working with DT, and PXA310/320/93x can be removed entirely. A good number of drivers become impossible to select after this, so each of these also get dropped. I'm including the driver patches in the series here and can either merge them through the soc tree, or they can get picked up by the individual subsystem maintainers. Since both the platform and the drivers get removed, the order should not matter. Note that the patches here are generated with 'git format-patch -D' and don't contain the contents of removed files because of the significant size (the first patch is close to a megabyte). Arnd Bergmann (30): ARM: pxa: remove unused board files ARM: pxa: remove irda leftover ARM: pxa: remove unused pxa3xx-ulpi ARM: pxa: drop pxa310/pxa320/pxa93x support ARM: pxa: prune unused device support power: remove z2_battery driver power: remove tosa_battery driver ata: remove palmld pata driver backlight: remove pxa tosa support input: remove pxa930_trkball driver input: remove pxa930_rotary keyboard driver input: remove zylonite touchscreen driver pcmcia: remove unused pxa/sa1100 drivers ASoC: pxa: remove unused board support ALSA: remove ac97 bus support mfd: remove wm97xx driver input: remove wm97xx support parport: remove ax88796 driver power: remove pda_power supply driver rtc: remove v3020 driver mfd: remove toshiba tmio drivers mfd: remove ucb1400 support mtd: remove tmio_nand driver mmc: remove tmio_mmc driver fbdev: remove tmiofb driver fbdev: remove w100fb driver leds: remove asic3 driver usb: remove ohci-tmio driver w1: remove ds1wm driver mfd: remove htc-pasic3 driver Cc: stern(a)rowland.harvard.edu Cc: alexandre.belloni(a)bootlin.com Cc: brgl(a)bgdev.pl Cc: damien.lemoal(a)opensource.wdc.com Cc: daniel(a)zonque.org Cc: dmitry.torokhov(a)gmail.com Cc: linux(a)dominikbrodowski.net Cc: balbi(a)kernel.org Cc: gregkh(a)linuxfoundation.org Cc: haojian.zhuang(a)gmail.com Cc: deller(a)gmx.de Cc: perex(a)perex.cz Cc: jingoohan1(a)gmail.com Cc: lee(a)kernel.org Cc: kernel(a)wantstofly.org Cc: lgirdwood(a)gmail.com Cc: linus.walleij(a)linaro.org Cc: marek.vasut(a)gmail.com Cc: broonie(a)kernel.org Cc: mkpetch(a)internode.on.net Cc: miquel.raynal(a)bootlin.com Cc: lost.distance(a)yahoo.com Cc: philipp.zabel(a)gmail.com Cc: robert.jarzmik(a)free.fr Cc: linux(a)armlinux.org.uk Cc: sre(a)kernel.org Cc: slapin(a)ossfans.org Cc: s.shtylyov(a)omp.ru Cc: sudipm.mukherjee(a)gmail.com Cc: tiwai(a)suse.com Cc: ulf.hansson(a)linaro.org Cc: vigneshr(a)ti.com Cc: viresh.kumar(a)linaro.org Cc: wsa+renesas(a)sang-engineering.com Cc: linux-pm(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-ide(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: patches(a)opensource.cirrus.com Cc: linux-leds(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-renesas-soc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-fbdev(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Documentation/power/power_supply_class.rst | 4 +- MAINTAINERS | 100 +- arch/arm/Kconfig | 1 - arch/arm/configs/cm_x300_defconfig | 163 -- arch/arm/configs/colibri_pxa270_defconfig | 157 -- arch/arm/configs/colibri_pxa300_defconfig | 60 - arch/arm/configs/corgi_defconfig | 247 --- arch/arm/configs/eseries_pxa_defconfig | 97 - arch/arm/configs/ezx_defconfig | 389 ---- arch/arm/configs/h5000_defconfig | 74 - arch/arm/configs/lpd270_defconfig | 58 - arch/arm/configs/lubbock_defconfig | 53 - arch/arm/configs/magician_defconfig | 151 -- arch/arm/configs/mainstone_defconfig | 51 - arch/arm/configs/palmz72_defconfig | 75 - arch/arm/configs/pcm027_defconfig | 90 - arch/arm/configs/pxa255-idp_defconfig | 55 - arch/arm/configs/pxa_defconfig | 3 - arch/arm/configs/trizeps4_defconfig | 207 --- arch/arm/configs/viper_defconfig | 160 -- arch/arm/configs/xcep_defconfig | 89 - arch/arm/configs/zeus_defconfig | 173 -- arch/arm/mach-pxa/Kconfig | 583 +----- arch/arm/mach-pxa/Makefile | 60 +- arch/arm/mach-pxa/balloon3-pcmcia.c | 137 -- arch/arm/mach-pxa/balloon3.c | 821 -------- arch/arm/mach-pxa/balloon3.h | 181 -- arch/arm/mach-pxa/capc7117.c | 159 -- arch/arm/mach-pxa/cm-x300.c | 883 --------- arch/arm/mach-pxa/colibri-evalboard.c | 138 -- arch/arm/mach-pxa/colibri-pcmcia.c | 165 -- arch/arm/mach-pxa/colibri-pxa270-income.c | 236 --- arch/arm/mach-pxa/colibri-pxa270.c | 330 ---- arch/arm/mach-pxa/colibri-pxa300.c | 193 -- arch/arm/mach-pxa/colibri-pxa320.c | 264 --- arch/arm/mach-pxa/colibri-pxa3xx.c | 147 -- arch/arm/mach-pxa/colibri.h | 70 - arch/arm/mach-pxa/corgi.c | 826 --------- arch/arm/mach-pxa/corgi.h | 110 -- arch/arm/mach-pxa/corgi_pm.c | 221 --- arch/arm/mach-pxa/csb701.c | 67 - arch/arm/mach-pxa/csb726.c | 291 --- arch/arm/mach-pxa/csb726.h | 24 - arch/arm/mach-pxa/devices.c | 408 ---- arch/arm/mach-pxa/devices.h | 3 - arch/arm/mach-pxa/e740-pcmcia.c | 127 -- arch/arm/mach-pxa/eseries-gpio.h | 63 - arch/arm/mach-pxa/eseries-irq.h | 24 - arch/arm/mach-pxa/eseries.c | 1001 ---------- arch/arm/mach-pxa/ezx.c | 1254 ------------- arch/arm/mach-pxa/h5000.c | 210 --- arch/arm/mach-pxa/h5000.h | 109 -- arch/arm/mach-pxa/himalaya.c | 166 -- arch/arm/mach-pxa/hx4700-pcmcia.c | 118 -- arch/arm/mach-pxa/hx4700.c | 942 ---------- arch/arm/mach-pxa/hx4700.h | 129 -- arch/arm/mach-pxa/icontrol.c | 218 --- arch/arm/mach-pxa/idp.c | 285 --- arch/arm/mach-pxa/idp.h | 195 -- arch/arm/mach-pxa/littleton.c | 462 ----- arch/arm/mach-pxa/littleton.h | 14 - arch/arm/mach-pxa/lpd270.c | 518 ------ arch/arm/mach-pxa/lpd270.h | 40 - arch/arm/mach-pxa/lubbock.c | 649 ------- arch/arm/mach-pxa/lubbock.h | 47 - arch/arm/mach-pxa/magician.c | 1112 ----------- arch/arm/mach-pxa/magician.h | 125 -- arch/arm/mach-pxa/mainstone.c | 738 -------- arch/arm/mach-pxa/mainstone.h | 140 -- arch/arm/mach-pxa/mfp-pxa25x.h | 33 - arch/arm/mach-pxa/mfp-pxa2xx.c | 4 - arch/arm/mach-pxa/mfp-pxa300.h | 52 - arch/arm/mach-pxa/mfp-pxa320.h | 458 ----- arch/arm/mach-pxa/mfp-pxa930.h | 495 ----- arch/arm/mach-pxa/mioa701.c | 784 -------- arch/arm/mach-pxa/mioa701.h | 76 - arch/arm/mach-pxa/mioa701_bootresume.S | 38 - arch/arm/mach-pxa/mp900.c | 101 - arch/arm/mach-pxa/mxm8x10.c | 477 ----- arch/arm/mach-pxa/mxm8x10.h | 22 - arch/arm/mach-pxa/palm27x.c | 473 ----- arch/arm/mach-pxa/palm27x.h | 77 - arch/arm/mach-pxa/palmld-pcmcia.c | 111 -- arch/arm/mach-pxa/palmld.c | 392 ---- arch/arm/mach-pxa/palmld.h | 107 -- arch/arm/mach-pxa/palmt5.c | 234 --- arch/arm/mach-pxa/palmt5.h | 82 - arch/arm/mach-pxa/palmtc-pcmcia.c | 162 -- arch/arm/mach-pxa/palmtc.c | 539 ------ arch/arm/mach-pxa/palmtc.h | 84 - arch/arm/mach-pxa/palmte2.c | 383 ---- arch/arm/mach-pxa/palmte2.h | 64 - arch/arm/mach-pxa/palmtreo.c | 548 ------ arch/arm/mach-pxa/palmtreo.h | 64 - arch/arm/mach-pxa/palmtx-pcmcia.c | 111 -- arch/arm/mach-pxa/palmtx.c | 390 ---- arch/arm/mach-pxa/palmtx.h | 110 -- arch/arm/mach-pxa/palmz72.c | 319 ---- arch/arm/mach-pxa/palmz72.h | 80 - arch/arm/mach-pxa/pcm027.c | 266 --- arch/arm/mach-pxa/pcm027.h | 73 - arch/arm/mach-pxa/pcm990-baseboard.c | 408 ---- arch/arm/mach-pxa/pcm990_baseboard.h | 199 -- arch/arm/mach-pxa/pm.c | 2 - arch/arm/mach-pxa/pm.h | 10 - arch/arm/mach-pxa/poodle.c | 484 ----- arch/arm/mach-pxa/poodle.h | 92 - arch/arm/mach-pxa/pxa25x.c | 7 - arch/arm/mach-pxa/pxa27x.c | 18 +- arch/arm/mach-pxa/pxa27x.h | 3 - arch/arm/mach-pxa/pxa2xx.c | 29 - arch/arm/mach-pxa/pxa300.c | 17 +- arch/arm/mach-pxa/pxa320.c | 88 - arch/arm/mach-pxa/pxa320.h | 9 - arch/arm/mach-pxa/pxa3xx-ulpi.c | 385 ---- arch/arm/mach-pxa/pxa3xx.c | 91 - arch/arm/mach-pxa/pxa930.c | 217 --- arch/arm/mach-pxa/pxa930.h | 8 - arch/arm/mach-pxa/pxa_cplds_irqs.c | 200 -- arch/arm/mach-pxa/regs-u2d.h | 199 -- arch/arm/mach-pxa/regs-uart.h | 146 -- arch/arm/mach-pxa/saar.c | 604 ------ arch/arm/mach-pxa/sharpsl_pm.c | 4 - arch/arm/mach-pxa/spitz.c | 23 - arch/arm/mach-pxa/tavorevb.c | 506 ----- arch/arm/mach-pxa/tosa-bt.c | 134 -- arch/arm/mach-pxa/tosa.c | 946 ---------- arch/arm/mach-pxa/tosa.h | 165 -- arch/arm/mach-pxa/tosa_bt.h | 18 - arch/arm/mach-pxa/trizeps4-pcmcia.c | 200 -- arch/arm/mach-pxa/trizeps4.c | 575 ------ arch/arm/mach-pxa/trizeps4.h | 166 -- arch/arm/mach-pxa/viper-pcmcia.c | 180 -- arch/arm/mach-pxa/viper-pcmcia.h | 12 - arch/arm/mach-pxa/viper.c | 1034 ----------- arch/arm/mach-pxa/viper.h | 91 - arch/arm/mach-pxa/vpac270-pcmcia.c | 137 -- arch/arm/mach-pxa/vpac270.c | 736 -------- arch/arm/mach-pxa/vpac270.h | 38 - arch/arm/mach-pxa/xcep.c | 190 -- arch/arm/mach-pxa/z2.c | 781 -------- arch/arm/mach-pxa/z2.h | 37 - arch/arm/mach-pxa/zeus.c | 974 ---------- arch/arm/mach-pxa/zeus.h | 82 - arch/arm/mach-pxa/zylonite.c | 495 ----- arch/arm/mach-pxa/zylonite.h | 45 - arch/arm/mach-pxa/zylonite_pxa300.c | 281 --- arch/arm/mach-pxa/zylonite_pxa320.c | 213 --- arch/arm/mach-tegra/tegra.c | 1 - drivers/ata/Kconfig | 9 - drivers/ata/Makefile | 1 - drivers/ata/pata_palmld.c | 137 -- drivers/clk/pxa/clk-pxa3xx.c | 22 +- drivers/cpufreq/pxa3xx-cpufreq.c | 21 +- drivers/gpio/Kconfig | 7 - drivers/gpio/Makefile | 1 - drivers/gpio/gpio-ucb1400.c | 85 - drivers/input/keyboard/Kconfig | 9 - drivers/input/keyboard/Makefile | 1 - drivers/input/keyboard/pxa930_rotary.c | 195 -- drivers/input/mouse/Kconfig | 6 - drivers/input/mouse/Makefile | 1 - drivers/input/mouse/pxa930_trkball.c | 250 --- drivers/input/touchscreen/Kconfig | 81 - drivers/input/touchscreen/Makefile | 8 - drivers/input/touchscreen/mainstone-wm97xx.c | 10 - drivers/input/touchscreen/ucb1400_ts.c | 458 ----- drivers/input/touchscreen/wm9705.c | 345 ---- drivers/input/touchscreen/wm9712.c | 466 ----- drivers/input/touchscreen/wm9713.c | 476 ----- drivers/input/touchscreen/wm97xx-core.c | 910 --------- drivers/input/touchscreen/zylonite-wm97xx.c | 220 --- drivers/leds/Kconfig | 11 - drivers/leds/Makefile | 1 - drivers/leds/leds-asic3.c | 177 -- drivers/mfd/Kconfig | 72 - drivers/mfd/Makefile | 8 - drivers/mfd/asic3.c | 1071 ----------- drivers/mfd/htc-pasic3.c | 210 --- drivers/mfd/t7l66xb.c | 432 ----- drivers/mfd/tc6387xb.c | 233 --- drivers/mfd/tc6393xb.c | 912 --------- drivers/mfd/tmio_core.c | 70 - drivers/mfd/ucb1400_core.c | 158 -- drivers/mfd/wm97xx-core.c | 362 ---- drivers/mmc/host/Kconfig | 8 - drivers/mmc/host/Makefile | 1 - drivers/mmc/host/pxamci.c | 3 +- drivers/mmc/host/tmio_mmc.c | 227 --- drivers/mtd/nand/raw/Kconfig | 7 - drivers/mtd/nand/raw/Makefile | 1 - drivers/mtd/nand/raw/tmio_nand.c | 533 ------ drivers/parport/Kconfig | 11 - drivers/parport/Makefile | 1 - drivers/parport/parport_ax88796.c | 418 ----- drivers/pcmcia/Kconfig | 12 +- drivers/pcmcia/Makefile | 5 - drivers/pcmcia/pxa2xx_base.c | 14 +- drivers/pcmcia/pxa2xx_mainstone.c | 122 -- drivers/pcmcia/sa1100_simpad.c | 115 -- drivers/pcmcia/sa1111_badge4.c | 158 -- drivers/pcmcia/sa1111_generic.c | 4 - drivers/pcmcia/sa1111_lubbock.c | 155 -- drivers/power/supply/Kconfig | 22 - drivers/power/supply/Makefile | 3 - drivers/power/supply/ds2760_battery.c | 8 - drivers/power/supply/pda_power.c | 520 ------ drivers/power/supply/tosa_battery.c | 512 ----- drivers/power/supply/z2_battery.c | 317 ---- drivers/rtc/Kconfig | 9 - drivers/rtc/Makefile | 1 - drivers/rtc/rtc-v3020.c | 369 ---- drivers/usb/gadget/udc/pxa25x_udc.c | 62 - drivers/usb/host/ohci-hcd.c | 18 - drivers/usb/host/ohci-pxa27x.c | 9 - drivers/usb/host/ohci-tmio.c | 364 ---- drivers/video/backlight/Kconfig | 14 - drivers/video/backlight/Makefile | 2 - drivers/video/backlight/locomolcd.c | 10 - drivers/video/backlight/tosa_bl.c | 173 -- drivers/video/backlight/tosa_bl.h | 8 - drivers/video/backlight/tosa_lcd.c | 284 --- drivers/video/fbdev/Kconfig | 39 - drivers/video/fbdev/Makefile | 2 - drivers/video/fbdev/tmiofb.c | 1040 ----------- drivers/video/fbdev/w100fb.c | 1644 ----------------- drivers/video/fbdev/w100fb.h | 924 --------- drivers/w1/masters/Kconfig | 7 - drivers/w1/masters/Makefile | 1 - drivers/w1/masters/ds1wm.c | 675 ------- include/linux/mfd/asic3.h | 313 ---- include/linux/mfd/htc-pasic3.h | 54 - include/linux/mfd/t7l66xb.h | 29 - include/linux/mfd/tc6387xb.h | 19 - include/linux/mfd/tc6393xb.h | 53 - include/linux/mfd/tmio.h | 5 - include/linux/parport.h | 5 - include/linux/pda_power.h | 39 - include/linux/platform_data/asoc-palm27x.h | 9 - include/linux/platform_data/asoc-poodle.h | 16 - include/linux/platform_data/asoc-pxa.h | 31 - include/linux/platform_data/irda-pxaficp.h | 26 - .../platform_data/keyboard-pxa930_rotary.h | 21 - include/linux/platform_data/mmp_audio.h | 18 - .../platform_data/mouse-pxa930_trkball.h | 11 - include/linux/platform_data/rtc-v3020.h | 41 - include/linux/platform_data/usb-pxa3xx-ulpi.h | 32 - include/linux/soc/pxa/cpu.h | 93 +- include/linux/ucb1400.h | 160 -- include/linux/z2_battery.h | 17 - include/sound/ac97/codec.h | 116 -- include/sound/ac97/compat.h | 17 - include/sound/ac97/controller.h | 83 - include/video/w100fb.h | 147 -- sound/Kconfig | 3 - sound/Makefile | 1 - sound/ac97/Kconfig | 19 - sound/ac97/Makefile | 9 - sound/ac97/ac97_core.h | 13 - sound/ac97/bus.c | 557 ------ sound/ac97/codec.c | 12 - sound/ac97/snd_ac97_compat.c | 120 -- sound/arm/Kconfig | 13 - sound/arm/Makefile | 4 - sound/arm/pxa2xx-ac97-lib.c | 466 ----- sound/arm/pxa2xx-ac97-regs.h | 100 - sound/arm/pxa2xx-ac97.c | 293 --- sound/pci/ac97/ac97_codec.c | 1 - sound/pci/ac97/ac97_patch.c | 40 - sound/soc/codecs/wm9705.c | 2 - sound/soc/codecs/wm9712.c | 2 - sound/soc/codecs/wm9713.c | 2 - sound/soc/pxa/Kconfig | 183 -- sound/soc/pxa/Makefile | 35 - sound/soc/pxa/brownstone.c | 133 -- sound/soc/pxa/corgi.c | 332 ---- sound/soc/pxa/e740_wm9705.c | 168 -- sound/soc/pxa/e750_wm9705.c | 147 -- sound/soc/pxa/e800_wm9712.c | 147 -- sound/soc/pxa/em-x270.c | 92 - sound/soc/pxa/hx4700.c | 207 --- sound/soc/pxa/magician.c | 366 ---- sound/soc/pxa/mioa701_wm9713.c | 201 -- sound/soc/pxa/mmp-pcm.c | 267 --- sound/soc/pxa/palm27x.c | 162 -- sound/soc/pxa/poodle.c | 291 --- sound/soc/pxa/pxa2xx-ac97.c | 307 --- sound/soc/pxa/pxa2xx-i2s.c | 2 - sound/soc/pxa/tosa.c | 255 --- sound/soc/pxa/ttc-dkb.c | 143 -- sound/soc/pxa/z2.c | 218 --- sound/soc/pxa/zylonite.c | 266 --- 292 files changed, 24 insertions(+), 59127 deletions(-) delete mode 100644 arch/arm/configs/cm_x300_defconfig delete mode 100644 arch/arm/configs/colibri_pxa270_defconfig delete mode 100644 arch/arm/configs/colibri_pxa300_defconfig delete mode 100644 arch/arm/configs/corgi_defconfig delete mode 100644 arch/arm/configs/eseries_pxa_defconfig delete mode 100644 arch/arm/configs/ezx_defconfig delete mode 100644 arch/arm/configs/h5000_defconfig delete mode 100644 arch/arm/configs/lpd270_defconfig delete mode 100644 arch/arm/configs/lubbock_defconfig delete mode 100644 arch/arm/configs/magician_defconfig delete mode 100644 arch/arm/configs/mainstone_defconfig delete mode 100644 arch/arm/configs/palmz72_defconfig delete mode 100644 arch/arm/configs/pcm027_defconfig delete mode 100644 arch/arm/configs/pxa255-idp_defconfig delete mode 100644 arch/arm/configs/trizeps4_defconfig delete mode 100644 arch/arm/configs/viper_defconfig delete mode 100644 arch/arm/configs/xcep_defconfig delete mode 100644 arch/arm/configs/zeus_defconfig delete mode 100644 arch/arm/mach-pxa/balloon3-pcmcia.c delete mode 100644 arch/arm/mach-pxa/balloon3.c delete mode 100644 arch/arm/mach-pxa/balloon3.h delete mode 100644 arch/arm/mach-pxa/capc7117.c delete mode 100644 arch/arm/mach-pxa/cm-x300.c delete mode 100644 arch/arm/mach-pxa/colibri-evalboard.c delete mode 100644 arch/arm/mach-pxa/colibri-pcmcia.c delete mode 100644 arch/arm/mach-pxa/colibri-pxa270-income.c delete mode 100644 arch/arm/mach-pxa/colibri-pxa270.c delete mode 100644 arch/arm/mach-pxa/colibri-pxa300.c delete mode 100644 arch/arm/mach-pxa/colibri-pxa320.c delete mode 100644 arch/arm/mach-pxa/colibri-pxa3xx.c delete mode 100644 arch/arm/mach-pxa/colibri.h delete mode 100644 arch/arm/mach-pxa/corgi.c delete mode 100644 arch/arm/mach-pxa/corgi.h delete mode 100644 arch/arm/mach-pxa/corgi_pm.c delete mode 100644 arch/arm/mach-pxa/csb701.c delete mode 100644 arch/arm/mach-pxa/csb726.c delete mode 100644 arch/arm/mach-pxa/csb726.h delete mode 100644 arch/arm/mach-pxa/e740-pcmcia.c delete mode 100644 arch/arm/mach-pxa/eseries-gpio.h delete mode 100644 arch/arm/mach-pxa/eseries-irq.h delete mode 100644 arch/arm/mach-pxa/eseries.c delete mode 100644 arch/arm/mach-pxa/ezx.c delete mode 100644 arch/arm/mach-pxa/h5000.c delete mode 100644 arch/arm/mach-pxa/h5000.h delete mode 100644 arch/arm/mach-pxa/himalaya.c delete mode 100644 arch/arm/mach-pxa/hx4700-pcmcia.c delete mode 100644 arch/arm/mach-pxa/hx4700.c delete mode 100644 arch/arm/mach-pxa/hx4700.h delete mode 100644 arch/arm/mach-pxa/icontrol.c delete mode 100644 arch/arm/mach-pxa/idp.c delete mode 100644 arch/arm/mach-pxa/idp.h delete mode 100644 arch/arm/mach-pxa/littleton.c delete mode 100644 arch/arm/mach-pxa/littleton.h delete mode 100644 arch/arm/mach-pxa/lpd270.c delete mode 100644 arch/arm/mach-pxa/lpd270.h delete mode 100644 arch/arm/mach-pxa/lubbock.c delete mode 100644 arch/arm/mach-pxa/lubbock.h delete mode 100644 arch/arm/mach-pxa/magician.c delete mode 100644 arch/arm/mach-pxa/magician.h delete mode 100644 arch/arm/mach-pxa/mainstone.c delete mode 100644 arch/arm/mach-pxa/mainstone.h delete mode 100644 arch/arm/mach-pxa/mfp-pxa320.h delete mode 100644 arch/arm/mach-pxa/mfp-pxa930.h delete mode 100644 arch/arm/mach-pxa/mioa701.c delete mode 100644 arch/arm/mach-pxa/mioa701.h delete mode 100644 arch/arm/mach-pxa/mioa701_bootresume.S delete mode 100644 arch/arm/mach-pxa/mp900.c delete mode 100644 arch/arm/mach-pxa/mxm8x10.c delete mode 100644 arch/arm/mach-pxa/mxm8x10.h delete mode 100644 arch/arm/mach-pxa/palm27x.c delete mode 100644 arch/arm/mach-pxa/palm27x.h delete mode 100644 arch/arm/mach-pxa/palmld-pcmcia.c delete mode 100644 arch/arm/mach-pxa/palmld.c delete mode 100644 arch/arm/mach-pxa/palmld.h delete mode 100644 arch/arm/mach-pxa/palmt5.c delete mode 100644 arch/arm/mach-pxa/palmt5.h delete mode 100644 arch/arm/mach-pxa/palmtc-pcmcia.c delete mode 100644 arch/arm/mach-pxa/palmtc.c delete mode 100644 arch/arm/mach-pxa/palmtc.h delete mode 100644 arch/arm/mach-pxa/palmte2.c delete mode 100644 arch/arm/mach-pxa/palmte2.h delete mode 100644 arch/arm/mach-pxa/palmtreo.c delete mode 100644 arch/arm/mach-pxa/palmtreo.h delete mode 100644 arch/arm/mach-pxa/palmtx-pcmcia.c delete mode 100644 arch/arm/mach-pxa/palmtx.c delete mode 100644 arch/arm/mach-pxa/palmtx.h delete mode 100644 arch/arm/mach-pxa/palmz72.c delete mode 100644 arch/arm/mach-pxa/palmz72.h delete mode 100644 arch/arm/mach-pxa/pcm027.c delete mode 100644 arch/arm/mach-pxa/pcm027.h delete mode 100644 arch/arm/mach-pxa/pcm990-baseboard.c delete mode 100644 arch/arm/mach-pxa/pcm990_baseboard.h delete mode 100644 arch/arm/mach-pxa/poodle.c delete mode 100644 arch/arm/mach-pxa/poodle.h delete mode 100644 arch/arm/mach-pxa/pxa320.c delete mode 100644 arch/arm/mach-pxa/pxa320.h delete mode 100644 arch/arm/mach-pxa/pxa3xx-ulpi.c delete mode 100644 arch/arm/mach-pxa/pxa930.c delete mode 100644 arch/arm/mach-pxa/pxa930.h delete mode 100644 arch/arm/mach-pxa/pxa_cplds_irqs.c delete mode 100644 arch/arm/mach-pxa/regs-u2d.h delete mode 100644 arch/arm/mach-pxa/regs-uart.h delete mode 100644 arch/arm/mach-pxa/saar.c delete mode 100644 arch/arm/mach-pxa/tavorevb.c delete mode 100644 arch/arm/mach-pxa/tosa-bt.c delete mode 100644 arch/arm/mach-pxa/tosa.c delete mode 100644 arch/arm/mach-pxa/tosa.h delete mode 100644 arch/arm/mach-pxa/tosa_bt.h delete mode 100644 arch/arm/mach-pxa/trizeps4-pcmcia.c delete mode 100644 arch/arm/mach-pxa/trizeps4.c delete mode 100644 arch/arm/mach-pxa/trizeps4.h delete mode 100644 arch/arm/mach-pxa/viper-pcmcia.c delete mode 100644 arch/arm/mach-pxa/viper-pcmcia.h delete mode 100644 arch/arm/mach-pxa/viper.c delete mode 100644 arch/arm/mach-pxa/viper.h delete mode 100644 arch/arm/mach-pxa/vpac270-pcmcia.c delete mode 100644 arch/arm/mach-pxa/vpac270.c delete mode 100644 arch/arm/mach-pxa/vpac270.h delete mode 100644 arch/arm/mach-pxa/xcep.c delete mode 100644 arch/arm/mach-pxa/z2.c delete mode 100644 arch/arm/mach-pxa/z2.h delete mode 100644 arch/arm/mach-pxa/zeus.c delete mode 100644 arch/arm/mach-pxa/zeus.h delete mode 100644 arch/arm/mach-pxa/zylonite.c delete mode 100644 arch/arm/mach-pxa/zylonite.h delete mode 100644 arch/arm/mach-pxa/zylonite_pxa300.c delete mode 100644 arch/arm/mach-pxa/zylonite_pxa320.c delete mode 100644 drivers/ata/pata_palmld.c delete mode 100644 drivers/gpio/gpio-ucb1400.c delete mode 100644 drivers/input/keyboard/pxa930_rotary.c delete mode 100644 drivers/input/mouse/pxa930_trkball.c delete mode 100644 drivers/input/touchscreen/ucb1400_ts.c delete mode 100644 drivers/input/touchscreen/wm9705.c delete mode 100644 drivers/input/touchscreen/wm9712.c delete mode 100644 drivers/input/touchscreen/wm9713.c delete mode 100644 drivers/input/touchscreen/wm97xx-core.c delete mode 100644 drivers/input/touchscreen/zylonite-wm97xx.c delete mode 100644 drivers/leds/leds-asic3.c delete mode 100644 drivers/mfd/asic3.c delete mode 100644 drivers/mfd/htc-pasic3.c delete mode 100644 drivers/mfd/t7l66xb.c delete mode 100644 drivers/mfd/tc6387xb.c delete mode 100644 drivers/mfd/tc6393xb.c delete mode 100644 drivers/mfd/tmio_core.c delete mode 100644 drivers/mfd/ucb1400_core.c delete mode 100644 drivers/mfd/wm97xx-core.c delete mode 100644 drivers/mmc/host/tmio_mmc.c delete mode 100644 drivers/mtd/nand/raw/tmio_nand.c delete mode 100644 drivers/parport/parport_ax88796.c delete mode 100644 drivers/pcmcia/pxa2xx_mainstone.c delete mode 100644 drivers/pcmcia/sa1100_simpad.c delete mode 100644 drivers/pcmcia/sa1111_badge4.c delete mode 100644 drivers/pcmcia/sa1111_lubbock.c delete mode 100644 drivers/power/supply/pda_power.c delete mode 100644 drivers/power/supply/tosa_battery.c delete mode 100644 drivers/power/supply/z2_battery.c delete mode 100644 drivers/rtc/rtc-v3020.c delete mode 100644 drivers/usb/host/ohci-tmio.c delete mode 100644 drivers/video/backlight/tosa_bl.c delete mode 100644 drivers/video/backlight/tosa_bl.h delete mode 100644 drivers/video/backlight/tosa_lcd.c delete mode 100644 drivers/video/fbdev/tmiofb.c delete mode 100644 drivers/video/fbdev/w100fb.c delete mode 100644 drivers/video/fbdev/w100fb.h delete mode 100644 drivers/w1/masters/ds1wm.c delete mode 100644 include/linux/mfd/asic3.h delete mode 100644 include/linux/mfd/htc-pasic3.h delete mode 100644 include/linux/mfd/t7l66xb.h delete mode 100644 include/linux/mfd/tc6387xb.h delete mode 100644 include/linux/mfd/tc6393xb.h delete mode 100644 include/linux/pda_power.h delete mode 100644 include/linux/platform_data/asoc-palm27x.h delete mode 100644 include/linux/platform_data/asoc-poodle.h delete mode 100644 include/linux/platform_data/asoc-pxa.h delete mode 100644 include/linux/platform_data/irda-pxaficp.h delete mode 100644 include/linux/platform_data/keyboard-pxa930_rotary.h delete mode 100644 include/linux/platform_data/mmp_audio.h delete mode 100644 include/linux/platform_data/mouse-pxa930_trkball.h delete mode 100644 include/linux/platform_data/rtc-v3020.h delete mode 100644 include/linux/platform_data/usb-pxa3xx-ulpi.h delete mode 100644 include/linux/ucb1400.h delete mode 100644 include/linux/z2_battery.h delete mode 100644 include/sound/ac97/codec.h delete mode 100644 include/sound/ac97/compat.h delete mode 100644 include/sound/ac97/controller.h delete mode 100644 include/video/w100fb.h delete mode 100644 sound/ac97/Kconfig delete mode 100644 sound/ac97/Makefile delete mode 100644 sound/ac97/ac97_core.h delete mode 100644 sound/ac97/bus.c delete mode 100644 sound/ac97/codec.c delete mode 100644 sound/ac97/snd_ac97_compat.c delete mode 100644 sound/arm/pxa2xx-ac97-lib.c delete mode 100644 sound/arm/pxa2xx-ac97-regs.h delete mode 100644 sound/arm/pxa2xx-ac97.c delete mode 100644 sound/soc/pxa/brownstone.c delete mode 100644 sound/soc/pxa/corgi.c delete mode 100644 sound/soc/pxa/e740_wm9705.c delete mode 100644 sound/soc/pxa/e750_wm9705.c delete mode 100644 sound/soc/pxa/e800_wm9712.c delete mode 100644 sound/soc/pxa/em-x270.c delete mode 100644 sound/soc/pxa/hx4700.c delete mode 100644 sound/soc/pxa/magician.c delete mode 100644 sound/soc/pxa/mioa701_wm9713.c delete mode 100644 sound/soc/pxa/mmp-pcm.c delete mode 100644 sound/soc/pxa/palm27x.c delete mode 100644 sound/soc/pxa/poodle.c delete mode 100644 sound/soc/pxa/pxa2xx-ac97.c delete mode 100644 sound/soc/pxa/tosa.c delete mode 100644 sound/soc/pxa/ttc-dkb.c delete mode 100644 sound/soc/pxa/z2.c delete mode 100644 sound/soc/pxa/zylonite.c -- 2.29.2

5 7

[PATCH] ALSA: usb-audio: fix Hamedal C20 usb camero disconnect issue
by aichao 04 Nov '22

04 Nov '22

For Hamedal C20, the current rate is different from the runtime rate, snd_usb_endpoint stop and close endpoint to resetting rate. if snd_usb_endpoint close the endpoint, sometimes usb will disconnect the device. Signed-off-by: aichao <aichao(a)kylinos.cn> --- sound/usb/endpoint.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 533919a28856..c19e420b1780 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -825,7 +825,9 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened) + if (!--ep->iface_ref->opened + && (le16_to_cpu(chip->dev->descriptor.idVendor) != 0x0525) + && (le16_to_cpu(chip->dev->descriptor.idProduct) != 0xa4ad)) endpoint_set_interface(chip, ep, false); if (!--ep->opened) { -- 2.25.1

3 2

[PATCH] ALSA: usb-audio: fix usb camero disconnect issue
by aichao 04 Nov '22

04 Nov '22

When current rate is different from the runtime rate, snd_usb_endpoint stop and close endpoint to resetting rate. if snd_usb_endpoint close the endpoint, sometimes usb will disconnect the device. Signed-off-by: aichao <aichao(a)kylinos.cn> --- sound/usb/pcm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5dc9266180e3..2ae4a55cbaa8 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -517,7 +517,6 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (snd_usb_endpoint_compatible(chip, subs->data_endpoint, fmt, hw_params)) goto unlock; - close_endpoints(chip, subs); } subs->data_endpoint = snd_usb_endpoint_open(chip, fmt, hw_params, false); @@ -546,9 +545,6 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, ret = configure_endpoints(chip, subs); unlock: - if (ret < 0) - close_endpoints(chip, subs); - snd_usb_unlock_shutdown(chip); stop_pipeline: if (ret < 0) @@ -574,7 +570,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) if (!snd_usb_lock_shutdown(chip)) { if (stop_endpoints(subs)) sync_pending_stops(subs); - close_endpoints(chip, subs); snd_usb_unlock_shutdown(chip); } -- 2.25.1

2 1

[PATCH 0/8] Adjust usage of rt5682(s) power supply properties
by Nícolas F. R. A. Prado 04 Nov '22

04 Nov '22

This series sets straight the usage of power supply properties for the rt5682 and rt5682s audio codecs. These properties were already being used by sc7180-trogdor.dtsi (and derived DTs like sc7180-trogdor-kingoftown.dtsi). We start by documenting the power supplies that are already in use and then add few others that were missing to the bindings. Then we update the drivers to also support the new supplies. Finally we update the trogdor DTs so they have the newly added but required supplies and remove a superfluous one that was causing warnings. Nícolas F. R. A. Prado (8): ASoC: dt-bindings: realtek,rt5682s: Add AVDD and MICVDD supplies ASoC: dt-bindings: realtek,rt5682s: Add dbvdd and ldo1-in supplies ASoC: dt-bindings: rt5682: Add AVDD, MICVDD and VBAT supplies ASoC: dt-bindings: rt5682: Add dbvdd and ldo1-in supplies ASoC: rt5682s: Support dbvdd and ldo1-in supplies ASoC: rt5682: Support dbvdd and ldo1-in supplies arm64: dts: qcom: sc7180-trogdor: Add missing supplies for rt5682 arm64: dts: qcom: sc7180-trogdor: Remove VBAT supply from rt5682s .../bindings/sound/realtek,rt5682s.yaml | 23 +++++++++++++++++++ .../devicetree/bindings/sound/rt5682.txt | 20 ++++++++++++++++ .../dts/qcom/sc7180-trogdor-kingoftown.dtsi | 1 + ...0-trogdor-wormdingler-rev1-boe-rt5682s.dts | 1 + ...0-trogdor-wormdingler-rev1-inx-rt5682s.dts | 1 + arch/arm64/boot/dts/qcom/sc7180-trogdor.dtsi | 2 ++ sound/soc/codecs/rt5682.c | 2 ++ sound/soc/codecs/rt5682.h | 2 +- sound/soc/codecs/rt5682s.c | 22 ++++++++++++++++++ sound/soc/codecs/rt5682s.h | 2 ++ 10 files changed, 75 insertions(+), 1 deletion(-) -- 2.38.1

5 18

[PATCH 0/2] drm: Add component_match_add_of and convert users of drm_of_component_match_add
by Sean Anderson 03 Nov '22

03 Nov '22

This series adds a new function component_match_add_of to simplify the common case of calling component_match_add_release with component_release_of and component_compare_of. There is already drm_of_component_match_add, which allows for a custom compare function. However, all existing users just use component_compare_of (or an equivalent). I can split the second commit up if that is easier to review. Sean Anderson (2): component: Add helper for device nodes drm: Convert users of drm_of_component_match_add to component_match_add_of .../gpu/drm/arm/display/komeda/komeda_drv.c | 6 ++-- drivers/gpu/drm/arm/hdlcd_drv.c | 9 +----- drivers/gpu/drm/arm/malidp_drv.c | 11 +------ drivers/gpu/drm/armada/armada_drv.c | 10 ++++--- drivers/gpu/drm/drm_of.c | 29 +++---------------- drivers/gpu/drm/etnaviv/etnaviv_drv.c | 4 +-- .../gpu/drm/hisilicon/kirin/kirin_drm_drv.c | 3 +- drivers/gpu/drm/ingenic/ingenic-drm-drv.c | 3 +- drivers/gpu/drm/mediatek/mtk_drm_drv.c | 4 +-- drivers/gpu/drm/msm/msm_drv.c | 14 ++++----- drivers/gpu/drm/sti/sti_drv.c | 3 +- drivers/gpu/drm/sun4i/sun4i_drv.c | 3 +- drivers/gpu/drm/tilcdc/tilcdc_external.c | 10 ++----- drivers/iommu/mtk_iommu.c | 3 +- drivers/iommu/mtk_iommu_v1.c | 3 +- include/drm/drm_of.h | 12 -------- include/linux/component.h | 9 ++++++ sound/soc/codecs/wcd938x.c | 6 ++-- 18 files changed, 46 insertions(+), 96 deletions(-) -- 2.35.1.1320.gc452695387.dirty

2 2

[PATCH] ASoC: simple-mux: add read function
by Maarten Zanders 03 Nov '22

03 Nov '22

During initialisation DAPM tries to read the state of the MUX being connected, resulting in this error log: input-mux: ASoC: error at soc_component_read_no_lock on input-mux: -5 Provide a read function which allows DAPM to read the state of the MUX. Signed-off-by: Maarten Zanders <maarten.zanders(a)mind.be> --- sound/soc/codecs/simple-mux.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/simple-mux.c b/sound/soc/codecs/simple-mux.c index d30c0d24d90a..bf67de12d20b 100644 --- a/sound/soc/codecs/simple-mux.c +++ b/sound/soc/codecs/simple-mux.c @@ -55,6 +55,14 @@ static int simple_mux_control_put(struct snd_kcontrol *kcontrol, e, NULL); } +static unsigned int simple_mux_read(struct snd_soc_component *component, + unsigned int reg) +{ + struct simple_mux *priv = snd_soc_component_get_drvdata(component); + + return priv->mux; +} + static const struct snd_kcontrol_new simple_mux_mux = SOC_DAPM_ENUM_EXT("Muxer", simple_mux_enum, simple_mux_control_get, simple_mux_control_put); @@ -76,6 +84,7 @@ static const struct snd_soc_component_driver simple_mux_component_driver = { .num_dapm_widgets = ARRAY_SIZE(simple_mux_dapm_widgets), .dapm_routes = simple_mux_dapm_routes, .num_dapm_routes = ARRAY_SIZE(simple_mux_dapm_routes), + .read = simple_mux_read, }; static int simple_mux_probe(struct platform_device *pdev) -- 2.37.3

2 1

[PATCH] Fix tas2562 volume table values
by Juan Alfonso Reyes Ajenjo 03 Nov '22

03 Nov '22

This patch fixes -90dB value (0x84a3 instead of 0x695b) This patch adds missing -20dB value (0x06666666) float_vol_db_lookup must have 56 values as volume is from 0 to 55 (0-110dB) Signed-off-by: Juan Alfonso Reyes Ajenjo jareyes(a)gmv.com<mailto:jareyes@gmv.com> Signed-off-by: Gustavo Plaza Roma gplaza(a)gmv.com<mailto:gplaza@gmv.com> --- sound/soc/codecs/tas2562.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b486d0b..acbdf77 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -33,15 +33,16 @@ static const unsigned int float_vol_db_lookup[] = { 0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, 0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, -0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x000084a3, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, 0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, 0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, 0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, 0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, 0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, 0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, -0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, -0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +0x06666666, 0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, +0x143d1362, 0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, +0x40000000 }; struct tas2562_data { -- 2.7.4

2 3

[PATCH] ASoC: Intel: cirrus-common: Make const array uid_strings static
by Colin Ian King 03 Nov '22

03 Nov '22

Don't populate the read-only const array uid_strings on the stack but instead make it static. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index 6e39eda77385..851c516c8f5b 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -155,7 +155,7 @@ static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; */ static int cs35l41_compute_codec_conf(void) { - const char * const uid_strings[] = { "0", "1", "2", "3" }; + static const char * const uid_strings[] = { "0", "1", "2", "3" }; unsigned int uid, sz = 0; struct acpi_device *adev; struct device *physdev; -- 2.38.1

3 2

[PATCH v3 1/2] ASoC: rt5514: fix legacy dai naming
by Jason Montleon 03 Nov '22

03 Nov '22

Starting with 6.0-rc1 these messages are logged and the sound card is unavailable. Adding legacy_dai_naming to the rt5514-spi causes it to function properly again. [ 16.928454] kbl_r5514_5663_max kbl_r5514_5663_max: ASoC: CPU DAI spi-PRP0001:00 not registered [ 16.928561] platform kbl_r5514_5663_max: deferred probe pending Signed-off-by: Jason Montleon <jmontleo(a)redhat.com> --- sound/soc/codecs/rt5514-spi.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 1a25a3787935..362663abcb89 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -298,13 +298,14 @@ static int rt5514_spi_pcm_new(struct snd_soc_component *component, } static const struct snd_soc_component_driver rt5514_spi_component = { - .name = DRV_NAME, - .probe = rt5514_spi_pcm_probe, - .open = rt5514_spi_pcm_open, - .hw_params = rt5514_spi_hw_params, - .hw_free = rt5514_spi_hw_free, - .pointer = rt5514_spi_pcm_pointer, - .pcm_construct = rt5514_spi_pcm_new, + .name = DRV_NAME, + .probe = rt5514_spi_pcm_probe, + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .pointer = rt5514_spi_pcm_pointer, + .pcm_construct = rt5514_spi_pcm_new, + .legacy_dai_naming = 1, }; /** -- 2.37.3

1 1

[PATCH v1 0/2] Revise mt8186 ADSP clock driver
by Tinghan Shen 03 Nov '22

03 Nov '22

Initialize the dependent clock sources for mt8186 ADSP and fix the enable/disable order of ADSP clocks. --- Tinghan Shen (2): dt-bindings: dsp: mediatek: Add default clock sources for mt8186 dsp ASoC: SOF: mediatek: Revise mt8186 ADSP clock driver .../bindings/dsp/mediatek,mt8186-dsp.yaml | 12 +++++-- sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 35 +++++++++++++++---- sound/soc/sof/mediatek/mt8186/mt8186-clk.h | 2 ++ 3 files changed, 39 insertions(+), 10 deletions(-) -- 2.18.0

3 4

building aplay / arecord
by David Jaffe 02 Nov '22

02 Nov '22

Newbie question... I downloaded the alsa-utils source and it says to run ./configure. But no such file exists. And if I go to aplay and type “make aplay” it can’t find a file called “aconfig.h”, which is presumably written by ./configure. What am I missing?

4 3

[PATCH v5 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 02 Nov '22

02 Nov '22

v5: - Documentation: - Add #include <dt-bindings/gpio/gpio.h> in order to use GPIO_ACTIVE_HIGH. - Remove the change-id in message. v4: - Documentation: - Fix the FATAL ERROR: Unable to parse input tree. v3: - Documentation: - Add an explain example in description. - Add the pinctrl-name and pinctrl id in its example. v2: - dmic codec driver: - Remove the unnecessary use of_property_read_bool() v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 14 ++- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 102 +++++++++++++++++- 2 files changed, 114 insertions(+), 2 deletions(-) -- 2.25.1

3 5

[PATCH linux-next] soundwire: qcom: remove redundant ret variable
by zhang.songyi＠zte.com.cn 02 Nov '22

02 Nov '22

>From ebc0be30e0c49d2d54a9aaa064c7f742e99e8eb2 Mon Sep 17 00:00:00 2001 From: zhang songyi <zhang.songyi(a)zte.com.cn> Date: Wed, 2 Nov 2022 22:08:48 +0800 Subject: [PATCH linux-next] soundwire: qcom: remove redundant ret variable Return value from swrm_get_packed_reg_val() directly instead of taking this in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: zhang songyi <zhang.songyi(a)zte.com.cn> --- drivers/soundwire/qcom.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 335424870290..d92f421d38c8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -260,7 +260,6 @@ static int qcom_swrm_cpu_reg_write(struct qcom_swrm_ctrl *ctrl, int reg, static u32 swrm_get_packed_reg_val(u8 *cmd_id, u8 cmd_data, u8 dev_addr, u16 reg_addr) { - u32 val; u8 id = *cmd_id; if (id != SWR_BROADCAST_CMD_ID) { @@ -270,9 +269,8 @@ static u32 swrm_get_packed_reg_val(u8 *cmd_id, u8 cmd_data, id = 0; *cmd_id = id; } - val = SWRM_REG_VAL_PACK(cmd_data, dev_addr, id, reg_addr); - return val; + return SWRM_REG_VAL_PACK(cmd_data, dev_addr, id, reg_addr); } static int swrm_wait_for_rd_fifo_avail(struct qcom_swrm_ctrl *swrm) -- 2.15.2

1 0

[PATCH] ALSA: usb-audio: Fix regression with Dell Dock jack detection
by Takashi Iwai 02 Nov '22

02 Nov '22

The recent commit added Jack controls to Dell Dock, but it added with iface = SNDRV_CTL_ELEM_IFACE_CARD. Unfortunately this doesn't match with the changes in user-space UCM profile, which expects iface = SNDRV_CTL_ELEM_IFACE_MIXER as default. This mismatch resulted in the non-working profile, and the Dell Dock is gone on pipewire / PulseAudio after the kernel update. Fix the regression by adjusting the iface of the new ctl elements to *_MIXER. Fixes: 4b8ea38fabab ("ALSA: usb-audio: Support jack detection on Dell dock") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.opensuse.org/show_bug.cgi?id=1204719 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ab0d459f4271..3039a17259ab 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2042,7 +2042,7 @@ static int realtek_ctl_connector_get(struct snd_kcontrol *kcontrol, } static const struct snd_kcontrol_new realtek_connector_ctl_ro = { - .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later manually */ .access = SNDRV_CTL_ELEM_ACCESS_READ, .info = snd_ctl_boolean_mono_info, -- 2.35.3

2 6

[PATCH v3 1/2] ASoC: Intel: Skylake: fix possible memory leak in skl_codec_device_init()
by Yang Yingliang 02 Nov '22

02 Nov '22

If snd_hdac_device_register() fails, 'codec' and name allocated in dev_set_name() called in snd_hdac_device_init() are leaked. Fix this by calling put_device(), so they can be freed in snd_hda_codec_dev_release() and kobject_cleanup(). Fixes: e4746d94d00c ("ASoC: Intel: Skylake: Introduce HDA codec init and exit routines") Fixes: dfe66a18780d ("ALSA: hdac_ext: add extended HDA bus") Suggested-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- v2 -> v3: add suggest tag v1 -> v2: remove skl_codec_device_exit() and use snd_hda_codec_dev_release() to fix leak --- sound/soc/intel/skylake/skl.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index bbba2df33aaf..3312b57e3c0c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -689,11 +689,6 @@ static void load_codec_module(struct hda_codec *codec) #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ -static void skl_codec_device_exit(struct device *dev) -{ - snd_hdac_device_exit(dev_to_hdac_dev(dev)); -} - static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) { struct hda_codec *codec; @@ -706,12 +701,11 @@ static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) } codec->core.type = HDA_DEV_ASOC; - codec->core.dev.release = skl_codec_device_exit; ret = snd_hdac_device_register(&codec->core); if (ret) { dev_err(bus->dev, "failed to register hdac device\n"); - snd_hdac_device_exit(&codec->core); + put_device(&codec->core.dev); return ERR_PTR(ret); } -- 2.25.1

3 3

[PATCH] ASoC: SOF: ipc3-topology: use old pipeline teardown flow with SOF2.1 and older
by Kai Vehmanen 02 Nov '22

02 Nov '22

Originally in commit b2ebcf42a48f ("ASoC: SOF: free widgets in sof_tear_down_pipelines() for static pipelines"), freeing of pipeline components at suspend was only done with recent FW as there were known limitations in older firmware versions. Tests show that if static pipelines are used, i.e. all pipelines are setup whenever firmware is powered up, the reverse action of freeing all components at power down, leads to firmware failures with also SOF2.0 and SOF2.1 based firmware. The problems can be specific to certain topologies with e.g. components not prepared to be freed at suspend (as this did not happen with older SOF kernels). To avoid hitting these problems when kernel is upgraded and used with an older firmware, bump the firmware requirement to SOF2.2 or newer. If an older firmware is used, and pipeline is a static one, do not free the components at suspend. This ensures the suspend flow remains backwards compatible with older firmware versions. This limitation does not apply if the product configuration is updated to dynamic pipelines. The limitation is not linked to firmware ABI, as the interface to free pipeline components has been available already before ABI3.19. The problem is in the implementation, so firmware version should be used to decide whether it is safe to use the newer flow or not. This patch adds a new SOF_FW_VER() macro to compare SOF firmware release versions. Link: https://github.com/thesofproject/sof/issues/6475 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- include/sound/sof/info.h | 4 ++++ sound/soc/sof/ipc3-topology.c | 15 ++++++++++----- 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index 65e86e4e9fd8..75193850ead0 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -36,6 +36,10 @@ enum sof_ipc_ext_data { SOF_IPC_EXT_USER_ABI_INFO = 4, }; +/* Build u32 number in format MMmmmppp */ +#define SOF_FW_VER(MAJOR, MINOR, PATCH) ((uint32_t)( \ + ((MAJOR) << 24) | ((MINOR) << 12) | (PATCH))) + /* FW version - SOF_IPC_GLB_VERSION */ struct sof_ipc_fw_version { struct sof_ipc_hdr hdr; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index c148715aa0f9..0720e1eae084 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2275,6 +2275,7 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; + bool dyn_widgets = false; int ret; /* @@ -2284,12 +2285,14 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif * topology loading the sound card unavailable to open PCMs. */ list_for_each_entry(swidget, &sdev->widget_list, list) { - if (swidget->dynamic_pipeline_widget) + if (swidget->dynamic_pipeline_widget) { + dyn_widgets = true; continue; + } - /* Do not free widgets for static pipelines with FW ABI older than 3.19 */ + /* Do not free widgets for static pipelines with FW older than SOF2.2 */ if (!verify && !swidget->dynamic_pipeline_widget && - v->abi_version < SOF_ABI_VER(3, 19, 0)) { + SOF_FW_VER(v->major, v->minor, v->micro) < SOF_FW_VER(2, 2, 0)) { swidget->use_count = 0; swidget->complete = 0; continue; @@ -2303,9 +2306,11 @@ static int sof_ipc3_tear_down_all_pipelines(struct snd_sof_dev *sdev, bool verif /* * Tear down all pipelines associated with PCMs that did not get suspended * and unset the prepare flag so that they can be set up again during resume. - * Skip this step for older firmware. + * Skip this step for older firmware unless topology has any + * dynamic pipeline (in which case the step is mandatory). */ - if (!verify && v->abi_version >= SOF_ABI_VER(3, 19, 0)) { + if (!verify && (dyn_widgets || SOF_FW_VER(v->major, v->minor, v->micro) >= + SOF_FW_VER(2, 2, 0))) { ret = sof_tear_down_left_over_pipelines(sdev); if (ret < 0) { dev_err(sdev->dev, "failed to tear down paused pipelines\n"); base-commit: b3d2170916491cbb849d08e5e816ec161947657d -- 2.38.1

2 1

Add support for getting list of supported codecs
by GitHub pull_request - edited 02 Nov '22

02 Nov '22

alsa-project/tinycompress pull request #16 was edited from SanchayanMaity: Right now, there is no way to expose a list of codecs from `tinycompress`. While one could theoretically call `is_codec_supported` multiple times in an application to check against a list of codecs, an API would make it easy to enumerate supported codecs. The IOCTL `SNDRV_COMPRESS_GET_CAPS` after all already exists. Our use case is Pipewire where we would like the compressed sink node to advertise only codecs supported by the underlying hardware. So far we were just advertising the complete list of codecs and let node linking fail at runtime based on result from `is_codec_supported`. Pipewire MR: https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/1431 Request URL : https://github.com/alsa-project/tinycompress/pull/16 Patch URL : https://github.com/alsa-project/tinycompress/pull/16.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ASoC: Intel: avs: fix possible memory leak in probe_codec()
by Yang Yingliang 02 Nov '22

02 Nov '22

If snd_hda_codec_configure() fails, 'codec' and name allocated in dev_set_name() called in snd_hdac_device_init() are leaked. Fix this by calling snd_hdac_device_unregister() and put_device(), so they can be freed in snd_hda_codec_dev_release() and kobject_cleanup(). Fixes: 1affc44ea5dd ("ASoC: Intel: avs: PCI driver implementation") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/intel/avs/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index bb0719c58ca4..2fde2b3f8790 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -140,6 +140,8 @@ static int probe_codec(struct hdac_bus *bus, int addr) ret = snd_hda_codec_configure(codec); if (ret < 0) { dev_err(bus->dev, "failed to config codec %d\n", ret); + snd_hdac_device_unregister(&codec->core); + put_device(&codec->core.dev); return ret; } -- 2.25.1

2 2

[PATCH 6.0 129/240] ASoC: codec: tlv320adc3xxx: add GPIOLIB dependency
by Greg Kroah-Hartman 02 Nov '22

02 Nov '22

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 551f2994b8ccdbe296e239278531e345d6e94d4d ] Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20221006235822.30074-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..932f08549788 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1599,6 +1599,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs. -- 2.35.1

1 0

[PATCH] ASOC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init()
by Yang Yingliang 02 Nov '22

02 Nov '22

If snd_hdac_device_register() fails, 'codec' is leaked, free it in snd_hdac_device_exit(). And device_initialize() is called in snd_hdac_device_init(), it should call put_device() to give up reference or the name allocated in dev_set_name() is leaked. The snd_hdac_device_exit() will be called in dev->release(). Fixes: 829c67319806 ("ASoC: SOF: Intel: Introduce HDA codec init and exit routines") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/intel/hda-codec.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 1e9afc48394c..f8cbcc25154b 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -111,7 +111,11 @@ EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); static void hda_codec_device_exit(struct device *dev) { - snd_hdac_device_exit(dev_to_hdac_dev(dev)); + struct hdac_device *hdac_dev = dev_to_hdac_dev(dev); + struct hda_codec *codec = container_of(hdac_dev, struct hda_codec, core); + + snd_hdac_device_exit(hdac_dev); + kfree(codec); } static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, int type) @@ -131,7 +135,7 @@ static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, i ret = snd_hdac_device_register(&codec->core); if (ret) { dev_err(bus->dev, "failed to register hdac device\n"); - snd_hdac_device_exit(&codec->core); + put_device(&codec->core.dev); return ERR_PTR(ret); } -- 2.25.1

2 2

[PATCH] ASoC: Intel: soc-acpi: add ES83x6 support to IceLake
by Pierre-Louis Bossart 01 Nov '22

01 Nov '22

Missing entry to find a machine driver for ES83x6-based platforms. Link: https://github.com/thesofproject/linux/issues/3873 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index b032bc07de8b..d0062f2cd256 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -10,6 +10,11 @@ #include <sound/soc-acpi-intel-match.h> #include "../skylake/skl.h" +static const struct snd_soc_acpi_codecs essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + static struct skl_machine_pdata icl_pdata = { .use_tplg_pcm = true, }; @@ -27,6 +32,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { .drv_name = "sof_rt5682", .sof_tplg_filename = "sof-icl-rt5682.tplg", }, + { + .comp_ids = &essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-icl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_icl_machines); -- 2.34.1

2 1

[PATCH] ASoC: hda: intel-dsp-config: add ES83x6 quirk for IceLake
by Pierre-Louis Bossart 01 Nov '22

01 Nov '22

Yet another hardware variant we need to handle. Link: https://github.com/thesofproject/linux/issues/3873 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9eb3208f288..ae31bb127594 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -320,6 +320,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x34c8, + .codec_hid = &essx_83x6, + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x34c8, -- 2.34.1

3 2

[PATCH 0/3] Add support for MICFIL on i.MX93 platform
by Chancel Liu 01 Nov '22

01 Nov '22

This patchset supports MICFIL on i.MX93 platform. Chancel Liu (3): ASoC: dt-bindings: fsl,micfil: Add compatible string for i.MX93 platform ASoC: fsl_micfil: Add support for i.MX93 platform ASoC: fsl_micfil: Add support when using eDMA .../devicetree/bindings/sound/fsl,micfil.yaml | 1 + sound/soc/fsl/fsl_micfil.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) -- 2.25.1

4 7

[PATCH v3] ASoC: cs42xx8-i2c.c: add module device table for of
by Peter Bergin 01 Nov '22

01 Nov '22

When trying to connect the device with the driver through device-tree it is not working. The of_device_id is defined in cs42xx8.c but is not correctly included in cs42xx8-i2c.c. Move of_device_id table to cs42xx8-i2c.c. Get cs42xx8_driver_data in cs42xx8_i2c_probe() and pass as argument to cs42xx8_probe(). Move error check if no driver data found to cs42xx8_i2c_probe(). Signed-off-by: Peter Bergin <peter(a)berginkonsult.se> --- v3: Revert new funtionality in _probe and not changing behavior. Move back data structs to common module to be able to use for other interfaces later. v2: reworked and removed duplication of cs42xx8_of_match sound/soc/codecs/cs42xx8-i2c.c | 28 +++++++++++++++++++++++++--- sound/soc/codecs/cs42xx8.c | 22 +++------------------- sound/soc/codecs/cs42xx8.h | 3 +-- 3 files changed, 29 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index bd80e9fc907f..052ffb7dcfc6 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -12,15 +12,30 @@ #include <linux/i2c.h> #include <linux/module.h> +#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <sound/soc.h> #include "cs42xx8.h" +static const struct of_device_id cs42xx8_of_match[]; + static int cs42xx8_i2c_probe(struct i2c_client *i2c) { - int ret = cs42xx8_probe(&i2c->dev, - devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + int ret; + struct cs42xx8_driver_data *drvdata; + const struct of_device_id *of_id; + + of_id = of_match_device(cs42xx8_of_match, &i2c->dev); + if (!of_id) { + dev_err(&i2c->dev, "failed to find driver data\n"); + return -EINVAL; + } + + drvdata = (struct cs42xx8_driver_data *)of_id->data; + + ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config), drvdata); if (ret) return ret; @@ -35,7 +50,14 @@ static void cs42xx8_i2c_remove(struct i2c_client *i2c) pm_runtime_disable(&i2c->dev); } -static struct i2c_device_id cs42xx8_i2c_id[] = { +static const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); + +static const struct i2c_device_id cs42xx8_i2c_id[] = { {"cs42448", (kernel_ulong_t)&cs42448_data}, {"cs42888", (kernel_ulong_t)&cs42888_data}, {} diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index d14eb2f6e1dd..4558ec38a7ac 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -13,7 +13,6 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/module.h> -#include <linux/of_device.h> #include <linux/gpio/consumer.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> @@ -511,17 +510,8 @@ const struct cs42xx8_driver_data cs42888_data = { }; EXPORT_SYMBOL_GPL(cs42888_data); -const struct of_device_id cs42xx8_of_match[] = { - { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, - { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, cs42xx8_of_match); -EXPORT_SYMBOL_GPL(cs42xx8_of_match); - -int cs42xx8_probe(struct device *dev, struct regmap *regmap) +int cs42xx8_probe(struct device *dev, struct regmap *regmap, struct cs42xx8_driver_data *drvdata) { - const struct of_device_id *of_id; struct cs42xx8_priv *cs42xx8; int ret, val, i; @@ -535,17 +525,11 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) if (cs42xx8 == NULL) return -ENOMEM; - cs42xx8->regmap = regmap; dev_set_drvdata(dev, cs42xx8); - of_id = of_match_device(cs42xx8_of_match, dev); - if (of_id) - cs42xx8->drvdata = of_id->data; + cs42xx8->regmap = regmap; - if (!cs42xx8->drvdata) { - dev_err(dev, "failed to find driver data\n"); - return -EINVAL; - } + cs42xx8->drvdata = drvdata; cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_HIGH); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h index d36c61b6df74..342389e8b1a8 100644 --- a/sound/soc/codecs/cs42xx8.h +++ b/sound/soc/codecs/cs42xx8.h @@ -22,8 +22,7 @@ extern const struct dev_pm_ops cs42xx8_pm; extern const struct cs42xx8_driver_data cs42448_data; extern const struct cs42xx8_driver_data cs42888_data; extern const struct regmap_config cs42xx8_regmap_config; -extern const struct of_device_id cs42xx8_of_match[]; -int cs42xx8_probe(struct device *dev, struct regmap *regmap); +int cs42xx8_probe(struct device *dev, struct regmap *regmap, struct cs42xx8_driver_data *drvdata); /* CS42888 register map */ #define CS42XX8_CHIPID 0x01 /* Chip ID */ -- 2.34.1

3 2

[PATCH] ASoC: SOF: fix compilation issue with readb/writeb helpers
by Pierre-Louis Bossart 01 Nov '22

01 Nov '22

Replace them with read8/write8 to avoid compilation issue on ARM. In hindsight this is more consistent with the read64/write64 helpers already used in SOF. Reported-by: Nathan Chancellor <nathan(a)kernel.org> Link: https://lore.kernel.org/alsa-devel/Y1rTFrohLqaiZAy%2F@dev-arch.thelio-3990X/ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 6 +++--- sound/soc/sof/intel/hda-stream.c | 2 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ops.h | 18 +++++++++--------- sound/soc/sof/sof-priv.h | 4 ++-- 5 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c61bab1a5719..6d5c26a2147e 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -349,7 +349,7 @@ static int hda_dsp_wait_d0i3c_done(struct snd_sof_dev *sdev) { int retry = HDA_DSP_REG_POLL_RETRY_COUNT; - while (snd_sof_dsp_readb(sdev, HDA_DSP_HDA_BAR, SOF_HDA_VS_D0I3C) & SOF_HDA_VS_D0I3C_CIP) { + while (snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, SOF_HDA_VS_D0I3C) & SOF_HDA_VS_D0I3C_CIP) { if (!retry--) return -ETIMEDOUT; usleep_range(10, 15); @@ -389,7 +389,7 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) } /* Update D0I3C register */ - snd_sof_dsp_updateb(sdev, HDA_DSP_HDA_BAR, + snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, SOF_HDA_VS_D0I3C, SOF_HDA_VS_D0I3C_I3, value); /* Wait for cmd in progress to be cleared before exiting the function */ @@ -399,7 +399,7 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) return ret; } - reg = snd_sof_dsp_readb(sdev, HDA_DSP_HDA_BAR, SOF_HDA_VS_D0I3C); + reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, SOF_HDA_VS_D0I3C); trace_sof_intel_D0I3C_updated(sdev, reg); return 0; diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 8e3f05857632..7f0fd05a96e6 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -455,7 +455,7 @@ int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, struct hdac_ext_st mask, mask); /* Follow HW recommendation to set the guardband value to 95us during FW boot */ - snd_sof_dsp_updateb(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_LTRP, + snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_LTRP, HDA_VS_INTEL_LTRP_GB_MASK, HDA_LTRP_GB_VALUE_US); /* start DMA */ diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 99be5ef78324..348fbfb6a2c2 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -630,7 +630,7 @@ void hda_ipc_irq_dump(struct snd_sof_dev *sdev) intsts = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS); intctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL); ppsts = snd_sof_dsp_read(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPSTS); - rirbsts = snd_sof_dsp_readb(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS); + rirbsts = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS); dev_err(sdev->dev, "hda irq intsts 0x%8.8x intlctl 0x%8.8x rirb %2.2x\n", intsts, intctl, rirbsts); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 8cb93e7c0c67..c52752250565 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -302,11 +302,11 @@ static inline int snd_sof_debugfs_add_region_item(struct snd_sof_dev *sdev, } /* register IO */ -static inline void snd_sof_dsp_writeb(struct snd_sof_dev *sdev, u32 bar, +static inline void snd_sof_dsp_write8(struct snd_sof_dev *sdev, u32 bar, u32 offset, u8 value) { - if (sof_ops(sdev)->writeb) - sof_ops(sdev)->writeb(sdev, sdev->bar[bar] + offset, value); + if (sof_ops(sdev)->write8) + sof_ops(sdev)->write8(sdev, sdev->bar[bar] + offset, value); else writeb(value, sdev->bar[bar] + offset); } @@ -329,11 +329,11 @@ static inline void snd_sof_dsp_write64(struct snd_sof_dev *sdev, u32 bar, writeq(value, sdev->bar[bar] + offset); } -static inline u8 snd_sof_dsp_readb(struct snd_sof_dev *sdev, u32 bar, +static inline u8 snd_sof_dsp_read8(struct snd_sof_dev *sdev, u32 bar, u32 offset) { - if (sof_ops(sdev)->readb) - return sof_ops(sdev)->readb(sdev, sdev->bar[bar] + offset); + if (sof_ops(sdev)->read8) + return sof_ops(sdev)->read8(sdev, sdev->bar[bar] + offset); else return readb(sdev->bar[bar] + offset); } @@ -356,15 +356,15 @@ static inline u64 snd_sof_dsp_read64(struct snd_sof_dev *sdev, u32 bar, return readq(sdev->bar[bar] + offset); } -static inline void snd_sof_dsp_updateb(struct snd_sof_dev *sdev, u32 bar, +static inline void snd_sof_dsp_update8(struct snd_sof_dev *sdev, u32 bar, u32 offset, u8 value, u8 mask) { u8 reg; - reg = snd_sof_dsp_readb(sdev, bar, offset); + reg = snd_sof_dsp_read8(sdev, bar, offset); reg &= ~mask; reg |= value; - snd_sof_dsp_writeb(sdev, bar, offset, reg); + snd_sof_dsp_write8(sdev, bar, offset, reg); } /* block IO */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 876e6fdbef4f..0ad48d73d1f7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -174,9 +174,9 @@ struct snd_sof_dsp_ops { * TODO: consider removing these operations and calling respective * implementations directly */ - void (*writeb)(struct snd_sof_dev *sof_dev, void __iomem *addr, + void (*write8)(struct snd_sof_dev *sof_dev, void __iomem *addr, u8 value); /* optional */ - u8 (*readb)(struct snd_sof_dev *sof_dev, + u8 (*read8)(struct snd_sof_dev *sof_dev, void __iomem *addr); /* optional */ void (*write)(struct snd_sof_dev *sof_dev, void __iomem *addr, u32 value); /* optional */ -- 2.34.1

2 1

[PATCH] ALSA: hda: clarify comments on SCF changes
by Pierre-Louis Bossart 01 Nov '22

01 Nov '22

The commit 1f9d3d98694b1 ("ALSA: hda - set intel audio clock to a proper value") added a number of misleading comments. There is no ability to detect if an SCF value was set or not, what the code does is prevent the use of the 6MHz audio clock represented by the value 0 in LCTL.SCF. Changing the SCF settings does require the link to be power-cycled, but in all other cases the link is powered automatically when exiting reset. In other words, the power-cycle is an exception to the rule that the HDaudio legacy driver does not need to program SPA/CPA bits. In addition, the SCF related changes are only relevant for the first link. No functionality change, only comment clarifications. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6ff19dd0d10c..92f66bbc5c91 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -485,8 +485,8 @@ static int intel_ml_lctl_set_power(struct azx *chip, int state) int timeout; /* - * the codecs are sharing the first link setting by default - * If other links are enabled for stream, they need similar fix + * Changes to LCTL.SCF are only needed for the first multi-link dealing + * with external codecs */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); val &= ~AZX_ML_LCTL_SPA; @@ -513,7 +513,7 @@ static void intel_init_lctl(struct azx *chip) /* 0. check lctl register value is correct or not */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); - /* if SCF is already set, let's use it */ + /* only perform additional configurations if the SCF is initially based on 6MHz */ if ((val & AZX_ML_LCTL_SCF) != 0) return; @@ -531,7 +531,7 @@ static void intel_init_lctl(struct azx *chip) if (ret) goto set_spa; - /* 2. update SCF to select a properly audio clock*/ + /* 2. update SCF to select an audio clock different from 6MHz */ val &= ~AZX_ML_LCTL_SCF; val |= intel_get_lctl_scf(chip); writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); -- 2.34.1

2 1

[PATCH v1 0/2] Add optional dmic selection for two DMICs
by Ajye Huang 01 Nov '22

01 Nov '22

v2: - Documentation: Add dmic_sel-gpios optional prop for two DMICs. - dmic codec driver: - "dmic_sel-gpios" property is used for a mixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: dt-bindings: Document dmic_sel-gpios optional prop for two DMICs case ASoC: dmic: Add optional dmic selection .../devicetree/bindings/sound/dmic-codec.yaml | 5 ++ sound/soc/codecs/dmic.c | 52 +++++++++++++++++++ 2 files changed, 57 insertions(+) -- 2.25.1

4 14

[PATCH] ASoC: soc-utils: Remove __exit for snd_soc_util_exit()
by Chen Zhongjin 31 Oct '22

31 Oct '22

snd_soc_util_exit() is called in __init snd_soc_init() for cleanup. Remove the __exit annotation for it to fix the build warning: WARNING: modpost: sound/soc/snd-soc-core.o: section mismatch in reference: init_module (section: .init.text) -> snd_soc_util_exit (section: .exit.text) Fixes: 6ec27c53886c ("ASoC: core: Fix use-after-free in snd_soc_exit()") Signed-off-by: Chen Zhongjin <chenzhongjin(a)huawei.com> --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index a3b6df2378b4..a4dba0b751e7 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -264,7 +264,7 @@ int __init snd_soc_util_init(void) return ret; } -void __exit snd_soc_util_exit(void) +void snd_soc_util_exit(void) { platform_driver_unregister(&soc_dummy_driver); platform_device_unregister(soc_dummy_dev); -- 2.17.1

2 1

[PATCH 0/5] ASoC: SOF: client-probes: Add support for IPC4
by Peter Ujfalusi 31 Oct '22

31 Oct '22

Hi, The probes (the ability of probing the audio data from firmware processing points or to receive stream of debug/trace information) is supported by IPC4 as well, but due to the differences between the two IPC version the low level setup and information we need for probing is different. This series will extend the existing probes support for IPC3 with IPC4 'backend' Regards, Peter --- Jyri Sarha (5): ASoC: SOF: probes: Replace [0] union members with DECLARE_FLEX_ARRAY() ASoC: SOF: probes: Separate IPC3 operations to a separate file ASoC: SOF: client: Add sof_client_ipc_set_get_data() ASoC: SOF: client: Add sof_client_ipc4_find_module() function ASoC: SOF: IPC4: probes: Implement IPC4 ops for probes client device sound/soc/sof/Makefile | 6 + sound/soc/sof/ipc.c | 8 + sound/soc/sof/sof-client-probes-ipc3.c | 236 +++++++++++++++++++++ sound/soc/sof/sof-client-probes-ipc4.c | 281 +++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 268 +++-------------------- sound/soc/sof/sof-client-probes.h | 34 +++ sound/soc/sof/sof-client.c | 34 +++ sound/soc/sof/sof-client.h | 6 + sound/soc/sof/sof-priv.h | 2 + 9 files changed, 637 insertions(+), 238 deletions(-) create mode 100644 sound/soc/sof/sof-client-probes-ipc3.c create mode 100644 sound/soc/sof/sof-client-probes-ipc4.c -- 2.38.1

2 6

[PATCH v1 0/3] ASoC: jz4740-i2s: Remove .set_sysclk() & friends
by Aidan MacDonald 31 Oct '22

31 Oct '22

A quick series to get rid of .set_sysclk() from jz4740-i2s. It wasn't used in practice so this shouldn't be troublesome for anyone, and fortunately there aren't any backward compatibility concerns. The actual rationale for removing it, as opposed to fixing the issues of the current DT bindings and implementation, is provided in the dt-bindings patch. Note the last patch only applies cleanly on top of my earlier jz4740-i2s cleanup series, but doesn't strictly depend on it. Link: https://lore.kernel.org/alsa-devel/20221023143328.160866-1-aidanmacdonald.0… Aidan MacDonald (3): dt-bindings: ingenic,aic: Remove unnecessary clocks from schema mips: dts: ingenic: Remove unnecessary AIC clocks ASoC: jz4740-i2s: Remove .set_sysclk() .../bindings/sound/ingenic,aic.yaml | 10 ++---- arch/mips/boot/dts/ingenic/jz4725b.dtsi | 7 ++-- arch/mips/boot/dts/ingenic/jz4740.dtsi | 7 ++-- arch/mips/boot/dts/ingenic/jz4770.dtsi | 5 ++- sound/soc/jz4740/jz4740-i2s.c | 32 ------------------- sound/soc/jz4740/jz4740-i2s.h | 10 ------ 6 files changed, 8 insertions(+), 63 deletions(-) delete mode 100644 sound/soc/jz4740/jz4740-i2s.h -- 2.38.1

4 8

[PATCH 1/2] slimbus: qcom-ngd-ctrl: check for device runtime PM status during ISR
by Krzysztof Kozlowski 31 Oct '22

31 Oct '22

Slimbus core interrupt is getting fired after suspend. At this point ADSP slimbus hardware is off with gated clocks which is leading to an unclocked access when HLOS slimbus tried to read the interrupt status register in the ISR. Co-developed-by: Chandana Kishori Chiluveru <cchiluve(a)codeaurora.org> Signed-off-by: Chandana Kishori Chiluveru <cchiluve(a)codeaurora.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index cec11aa106bf..ba36eb5c0de3 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -763,7 +763,14 @@ static irqreturn_t qcom_slim_ngd_interrupt(int irq, void *d) { struct qcom_slim_ngd_ctrl *ctrl = d; void __iomem *base = ctrl->ngd->base; - u32 stat = readl(base + NGD_INT_STAT); + u32 stat; + + if (pm_runtime_suspended(ctrl->ctrl.dev)) { + dev_warn_once(ctrl->dev, "Interrupt received while suspended\n"); + return IRQ_NONE; + } + + stat = readl(base + NGD_INT_STAT); if ((stat & NGD_INT_MSG_BUF_CONTE) || (stat & NGD_INT_MSG_TX_INVAL) || (stat & NGD_INT_DEV_ERR) || -- 2.34.1

2 2

[PATCH v2 1/2] slimbus: stream: handle unsupported bitrates for presence rate
by Krzysztof Kozlowski 31 Oct '22

31 Oct '22

Handle errors of getting presence rate for unsupported stream bitrate, instead of sending -EINVAL in change content message. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Konrad Dybcio <konrad.dybcio(a)somainline.org> --- Changes since v1: 1. Rebase (drop development pieces in the context). 2. Return prrate (which has errno) instead of -EINVAL. 3. Add Rb tag. --- drivers/slimbus/stream.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index d5ce974e880b..f631d5ed9809 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -204,7 +204,7 @@ int slim_stream_prepare(struct slim_stream_runtime *rt, { struct slim_controller *ctrl = rt->dev->ctrl; struct slim_port *port; - int num_ports, i, port_id; + int num_ports, i, port_id, prrate; if (rt->ports) { dev_err(&rt->dev->dev, "Stream already Prepared\n"); @@ -221,6 +221,13 @@ int slim_stream_prepare(struct slim_stream_runtime *rt, rt->bps = cfg->bps; rt->direction = cfg->direction; + prrate = slim_get_prate_code(cfg->rate); + if (prrate < 0) { + dev_err(&rt->dev->dev, "Cannot get presence rate for rate %d Hz\n", + cfg->rate); + return prrate; + } + if (cfg->rate % ctrl->a_framer->superfreq) { /* * data rate not exactly multiple of super frame, @@ -241,7 +248,7 @@ int slim_stream_prepare(struct slim_stream_runtime *rt, port = &rt->ports[i]; port->state = SLIM_PORT_DISCONNECTED; port->id = port_id; - port->ch.prrate = slim_get_prate_code(cfg->rate); + port->ch.prrate = prrate; port->ch.id = cfg->chs[i]; port->ch.data_fmt = SLIM_CH_DATA_FMT_NOT_DEFINED; port->ch.aux_fmt = SLIM_CH_AUX_FMT_NOT_APPLICABLE; -- 2.34.1

2 2

[PATCH] slimbus: stream: correct presence rate frequencies
by Krzysztof Kozlowski 31 Oct '22

31 Oct '22

Correct few frequencies in presence rate table - multiplied by 10 (110250 instead of 11025 Hz). Fixes: abb9c9b8b51b ("slimbus: stream: add stream support") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index f7da72ab504c..a476fa6549cc 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -67,10 +67,10 @@ static const int slim_presence_rate_table[] = { 384000, 768000, 0, /* Reserved */ - 110250, - 220500, - 441000, - 882000, + 11025, + 22050, + 44100, + 88200, 176400, 352800, 705600, -- 2.34.1

2 1

[PATCH -next] slimbus: qcom-ngd: Fix build error when CONFIG_SLIM_QCOM_NGD_CTRL=y && CONFIG_QCOM_RPROC_COMMON=m
by Zheng Bin 31 Oct '22

31 Oct '22

If CONFIG_SLIM_QCOM_NGD_CTRL=y, CONFIG_QCOM_RPROC_COMMON=m, COMPILE_TEST=y, bulding fails: drivers/slimbus/qcom-ngd-ctrl.o: In function `qcom_slim_ngd_ctrl_probe': qcom-ngd-ctrl.c:(.text+0x330): undefined reference to `qcom_register_ssr_notifier' qcom-ngd-ctrl.c:(.text+0x5fc): undefined reference to `qcom_unregister_ssr_notifier' drivers/slimbus/qcom-ngd-ctrl.o: In function `qcom_slim_ngd_remove': qcom-ngd-ctrl.c:(.text+0x90c): undefined reference to `qcom_unregister_ssr_notifier' Make SLIM_QCOM_NGD_CTRL depends on QCOM_RPROC_COMMON || (COMPILE_TEST && !QCOM_RPROC_COMMON) to fix this. Fixes: e291691c6977 ("slimbus: qcom-ngd-ctrl: allow compile testing without QCOM_RPROC_COMMON") Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> --- drivers/slimbus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/Kconfig b/drivers/slimbus/Kconfig index 2ed821f75816..a0fdf9d792cb 100644 --- a/drivers/slimbus/Kconfig +++ b/drivers/slimbus/Kconfig @@ -23,7 +23,7 @@ config SLIM_QCOM_CTRL config SLIM_QCOM_NGD_CTRL tristate "Qualcomm SLIMbus Satellite Non-Generic Device Component" depends on HAS_IOMEM && DMA_ENGINE && NET - depends on QCOM_RPROC_COMMON || COMPILE_TEST + depends on QCOM_RPROC_COMMON || (COMPILE_TEST && !QCOM_RPROC_COMMON) depends on ARCH_QCOM || COMPILE_TEST select QCOM_QMI_HELPERS select QCOM_PDR_HELPERS -- 2.31.1

3 2

[PATCH 1/5] slimbus: stream: add checks for invalid unprepare/disable usage
by Krzysztof Kozlowski 31 Oct '22

31 Oct '22

slim_disable_stream() and slim_stream_unprepare() are exported, so add sanity checks preventing unmatched/invalid calls. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/slimbus/stream.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 75f87b3d8b95..7bb073ca6006 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -407,6 +407,9 @@ int slim_stream_disable(struct slim_stream_runtime *stream) struct slim_controller *ctrl = stream->dev->ctrl; int ret, i; + if (!stream->ports || !stream->num_ports) + return -EINVAL; + if (ctrl->disable_stream) ctrl->disable_stream(stream); @@ -438,6 +441,9 @@ int slim_stream_unprepare(struct slim_stream_runtime *stream) { int i; + if (!stream->ports || !stream->num_ports) + return -EINVAL; + for (i = 0; i < stream->num_ports; i++) slim_disconnect_port(stream, &stream->ports[i]); -- 2.34.1

3 10

feat: add support for ASUS ROG Strix Z790-E Gaming Wifi
by GitHub pull_request - opened 31 Oct '22

31 Oct '22

alsa-project/alsa-ucm-conf pull request #235 was opened from DASPRiD: I've verified this profile on my motherboard, and both back- and front jacks seem to work fine, including SPDIF. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/235 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/235.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/3] ASoC: tas2770: Fix set_tdm_slot in case of single slot
by Martin Povišer 31 Oct '22

31 Oct '22

There's a special branch in the set_tdm_slot op for the case of nslots being 1, but: (1) That branch can never work (there's a check for tx_mask being non-zero, later there's another check for it *being* zero; one or the other always throws -EINVAL). (2) The intention of the branch seems to be what the general other branch reduces to in case of nslots being 1. For those reasons remove the 'nslots being 1' special case. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Suggested-by: Jos Dehaes <jos.dehaes(a)gmail.com> Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/codecs/tas2770.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index b6765235a4b3..8557759acb1f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -395,21 +395,13 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, if (tx_mask == 0 || rx_mask != 0) return -EINVAL; - if (slots == 1) { - if (tx_mask != 1) - return -EINVAL; - - left_slot = 0; - right_slot = 0; + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; } else { - left_slot = __ffs(tx_mask); - tx_mask &= ~(1 << left_slot); - if (tx_mask == 0) { - right_slot = left_slot; - } else { - right_slot = __ffs(tx_mask); - tx_mask &= ~(1 << right_slot); - } + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) -- 2.33.0

2 3

[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

11 28

[PATCH] ASoC: core: Fix use-after-free in snd_soc_exit()
by Chen Zhongjin 31 Oct '22

31 Oct '22

KASAN reports a use-after-free: BUG: KASAN: use-after-free in device_del+0xb5b/0xc60 Read of size 8 at addr ffff888008655050 by task rmmod/387 CPU: 2 PID: 387 Comm: rmmod Hardware name: QEMU Standard PC (i440FX + PIIX, 1996) Call Trace: <TASK> dump_stack_lvl+0x79/0x9a print_report+0x17f/0x47b kasan_report+0xbb/0xf0 device_del+0xb5b/0xc60 platform_device_del.part.0+0x24/0x200 platform_device_unregister+0x2e/0x40 snd_soc_exit+0xa/0x22 [snd_soc_core] __do_sys_delete_module.constprop.0+0x34f/0x5b0 do_syscall_64+0x3a/0x90 entry_SYSCALL_64_after_hwframe+0x63/0xcd ... </TASK> It's bacause in snd_soc_init(), snd_soc_util_init() is possble to fail, but its ret is ignored, which makes soc_dummy_dev unregistered twice. snd_soc_init() snd_soc_util_init() platform_device_register_simple(soc_dummy_dev) platform_driver_register() # fail platform_device_unregister(soc_dummy_dev) platform_driver_register() # success ... snd_soc_exit() snd_soc_util_exit() # soc_dummy_dev will be unregistered for second time To fix it, handle error and stop snd_soc_init() when util_init() fail. Also clean debugfs when util_init() or driver_register() fail. Fixes: fb257897bf20 ("ASoC: Work around allmodconfig failure") Signed-off-by: Chen Zhongjin <chenzhongjin(a)huawei.com> --- sound/soc/soc-core.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 12a82f5a3ff6..a409fbed8f34 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3477,10 +3477,23 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_cpus); static int __init snd_soc_init(void) { + int ret; + snd_soc_debugfs_init(); - snd_soc_util_init(); + ret = snd_soc_util_init(); + if (ret) + goto err_util_init; - return platform_driver_register(&soc_driver); + ret = platform_driver_register(&soc_driver); + if (ret) + goto err_register; + return 0; + +err_register: + snd_soc_util_exit(); +err_util_init: + snd_soc_debugfs_exit(); + return ret; } module_init(snd_soc_init); -- 2.17.1

2 4

[PATCH v4 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 31 Oct '22

31 Oct '22

v4: - Documentation: - Fix the FATAL ERROR: Unable to parse input tree. v3: - Documentation: - Add an explain example in description. - Add the pinctrl-name and pinctrl id in its example. v2: - dmic codec driver: - Remove the unnecessary use of_property_read_bool() v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 13 ++- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 102 +++++++++++++++++- 2 files changed, 113 insertions(+), 2 deletions(-) -- 2.25.1

1 3

Missing USB IDs for M-Audio Micro
by Ash Logan 31 Oct '22

31 Oct '22

Hello! I have an M-Audio Micro USB soundcard, which works great, though I have to manually add the USB IDs to snd-usb-audio when I want to use it. echo 0763 201a > /sys/module/snd_usb_audio/drivers/usb\:snd-usb-audio/new_id I was hoping you could add the IDs somewhere in the driver to enable it by default, since the output, input and even bass boost "feature" seem to work just fine. Below is the lsusb information for this device. Thanks! Ash Bus 003 Device 004: ID 0763:201a M-Audio M-Audio Micro Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0763 M-Audio idProduct 0x201a M-Audio Micro bcdDevice 0.08 iManufacturer 1 M-Audio iProduct 2 M-Audio Micro iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0186 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 3 G7 2006/06/08 16:08 bmAttributes 0x80 (Bus Powered) MaxPower 100mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x007c bInCollection 2 baInterfaceNr(0) 1 baInterfaceNr(1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 10 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 11 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 10 bControlSize 2 bmaControls(0) 0x0002 Volume Control bmaControls(1) 0x0000 bmaControls(2) 0x0000 iFeature 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 3 bSourceID 11 bControlSize 2 bmaControls(0) 0x0002 Volume Control bmaControls(1) 0x0000 bmaControls(2) 0x0000 iFeature 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 2 baSourceID(0) 2 baSourceID(1) 3 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 1 bSourceID 12 bControlSize 2 bmaControls(0) 0x0155 Mute Control Bass Control Treble Control Automatic Gain Control Bass Boost Control bmaControls(1) 0x0002 Volume Control bmaControls(2) 0x0002 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 20 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 7 bControlSize 2 bmaControls(0) 0x0003 Mute Control Volume Control bmaControls(1) 0x0000 bmaControls(2) 0x0000 iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 13 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 6 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 12 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 6400 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0001 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 12 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 6400 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0001 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 13 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 1 bBitResolution 8 bSamFreqType 0 Continuous tLowerSamFreq 6400 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0032 1x 50 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0001 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 13 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 6400 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0001 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 13 bDelay 0 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 6400 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 2 Decoded PCM samples wLockDelay 0x0001 can't get debug descriptor: Resource temporarily unavailable Device Status: 0x0000 (Bus Powered)

2 1

utils/alsa.m4: include <stdlib.h> for exit()
by GitHub pull_request - opened 30 Oct '22

30 Oct '22

alsa-project/alsa-lib pull request #278 was opened from thesamesam: Clang 16 makes -Wimplicit-function-declaration an error by default and it's easier to just explicitly include <stdlib.h> rather than hope the ALSA headers always have it. Signed-off-by: Sam James <sam(a)gentoo.org> Request URL : https://github.com/alsa-project/alsa-lib/pull/278 Patch URL : https://github.com/alsa-project/alsa-lib/pull/278.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 30 Oct '22

30 Oct '22

v3: - Documentation: - Add an explain example in description. - Add the pinctrl-name and pinctrl id in its example. v2: - dmic codec driver: - Remove the unnecessary use of_property_read_bool() v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 13 ++- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 102 +++++++++++++++++- 2 files changed, 113 insertions(+), 2 deletions(-) -- 2.25.1

2 4

wave: add <time.h> missing header inclusion
by GitHub pull_request - opened 29 Oct '22

29 Oct '22

alsa-project/tinycompress pull request #19 was opened from jolivain: When compiling tinycompress with some libc other than glibc (e.g. musl libc), compilation fails with the following error: In file included from wave.c:11:0: /path/to/gcc/sysroot/usr/include/sound/asound.h:404:18: error: field 'trigger_tstamp' has incomplete type struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ [...] According to POSIX, `struct timespec` is defined in `<time.h>`. See: https://pubs.opengroup.org/onlinepubs/9699919799/basedefs/time.h.html This patch fixes this build failure by including `<time.h>` prior `<sound/asound.h>`. Note: the github master branch is currently not including the tag 1.2.8, which is why my pull-request is also showing commit 750ebbe. Request URL : https://github.com/alsa-project/tinycompress/pull/19 Patch URL : https://github.com/alsa-project/tinycompress/pull/19.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH RESEND] ARM: pxa: pxa2xx-ac97-lib: fix return value check of platform_get_irq()
by Yang Yingliang 29 Oct '22

29 Oct '22

platform_get_irq() returns negative error number on failure, fix the return value check in pxa2xx_ac97_hw_probe() and assign the error code to 'ret'. Fixes: 2548e6c76ebf ("ARM: pxa: pxa2xx-ac97-lib: use IRQ resource") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- The previous link: https://lore.kernel.org/lkml/20220425074559.2436002-1-yangyingliang@huawei.… --- sound/arm/pxa2xx-ac97-lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e55c0421718b..2ca33fd5a575 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -402,8 +402,10 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk2; irq = platform_get_irq(dev, 0); - if (!irq) + if (irq < 0) { + ret = irq; goto err_irq; + } ret = request_irq(irq, pxa2xx_ac97_irq, 0, "AC97", NULL); if (ret < 0) -- 2.25.1

2 1

[PATCH v2 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 29 Oct '22

29 Oct '22

v2: - dmic codec driver: - Remove the unnecessary use of_property_read_bool() v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 6 ++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 102 +++++++++++++++++- 2 files changed, 107 insertions(+), 1 deletion(-) -- 2.25.1

3 7

[PATCH v2] ASoC: sun50i-dmic: avoid unused variable warning for sun50i_dmic_of_match
by Ban Tao 29 Oct '22

29 Oct '22

In configurations with CONFIG_OF=n, we get a harmless build warning: sound/soc/sunxi/sun50i-dmic.c:268:34: warning: unused variable 'sun50i_dmic_of_match' [-Wunused-const-variable] Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reported-by: kernel test robot <lkp(a)intel.com> Acked-by: Jernej Skrabec <jernej.skrabec(a)gmail.com> --- v1->v2:Add "Acked-by" tag. --- sound/soc/sunxi/sun50i-dmic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 86cff5a..62509cb 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -391,7 +391,7 @@ static const struct dev_pm_ops sun50i_dmic_pm = { static struct platform_driver sun50i_dmic_driver = { .driver = { .name = "sun50i-dmic", - .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .of_match_table = sun50i_dmic_of_match, .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, -- 2.7.4

4 3

[PATCH 00/12] ASoC: mediatek: Add support for MT8188 SoC
by Trevor Wu 29 Oct '22

29 Oct '22

This series of patches adds support for Mediatek AFE of MT8188 SoC. Patches are based on broonie tree "for-next" branch. Trevor Wu (12): ASoC: mediatek: common: add SMC ops ID ASoC: mediatek: mt8188: add common header ASoC: mediatek: mt8188: support audsys clock ASoC: mediatek: mt8188: support adda in platform driver ASoC: mediatek: mt8188: support etdm in platform driver ASoC: mediatek: mt8188: support pcmif in platform driver ASoC: mediatek: mt8188: support audio clock control ASoC: mediatek: mt8188: add platform driver ASoC: mediatek: mt8188: add control for timing select dt-bindings: mediatek: mt8188: add audio afe document ASoC: mediatek: mt8188: add machine driver with mt6359 dt-bindings: mediatek: mt8188: add mt8188-mt6359 document .../bindings/sound/mt8188-afe-pcm.yaml | 202 + .../bindings/sound/mt8188-mt6359.yaml | 70 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-base-afe.h | 16 + sound/soc/mediatek/mt8188/Makefile | 15 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 656 ++++ sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 114 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 151 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 3451 +++++++++++++++++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 206 + sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 15 + .../soc/mediatek/mt8188/mt8188-audsys-clkid.h | 83 + sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 639 +++ sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 2598 +++++++++++++ sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 362 ++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 810 ++++ sound/soc/mediatek/mt8188/mt8188-reg.h | 3300 ++++++++++++++++ 18 files changed, 12712 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8188-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8188-mt6359.yaml create mode 100644 sound/soc/mediatek/mt8188/Makefile create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-common.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-adda.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-mt6359.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-reg.h -- 2.18.0

5 26

[PATCH v2 0/4] Add missing dt-binding properties to rt5682(s)
by Nícolas F. R. A. Prado 29 Oct '22

29 Oct '22

This series adds the missing sound-dai-cells to rt5682s and supplies for both rt5682s and rt5682. These properties are already used by sc7180-trogdor.dtsi (and derived DTs like sc7180-trogdor-kingoftown.dtsi), so the commits in this series are really just documenting their usage. v1: https://lore.kernel.org/all/20221021190908.1502026-1-nfraprado@collabora.co… Changes in v2: - Changed sound-dai-cells to 1 on rt5682s - Added commit fixing sound-dai-cells on rt5682 - Added mention to commit messages that properties are already being used by DTs Nícolas F. R. A. Prado (4): ASoC: dt-bindings: realtek,rt5682s: Add #sound-dai-cells ASoC: dt-bindings: realtek,rt5682s: Add AVDD and MICVDD supplies ASoC: dt-bindings: rt5682: Set sound-dai-cells to 1 ASoC: dt-bindings: rt5682: Add AVDD, MICVDD and VBAT supplies .../devicetree/bindings/sound/realtek,rt5682s.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/rt5682.txt | 8 +++++++- 2 files changed, 14 insertions(+), 1 deletion(-) -- 2.38.1

5 23

[PATCH v3 0/9] ASoC: qdsp6: audioreach: add multi-port, SAL and MFC support
by Srinivas Kandagatla 28 Oct '22

28 Oct '22

This patchset adds support to multi-port connections between AudioReach Modules which is required for sophisticated graphs like ECNS or Speaker Protection. Also as part of ECNS testing new module support for SAL and MFC are added. Tested on SM8450 with ECNS. Thanks, Srini Changes since v2: Fixed two spelling mistakes in commit message No code changes. Srinivas Kandagatla (9): ASoC: qdsp6: audioreach: topology use idr_alloc_u32 ASoC: qdsp6: audioreach: remove unused connection_list ASoC: qdsp6: audioreach: update dapm kcontrol private data ASoC: qdsp6: audioreach: Simplify handing FE and BE graph connections ASoC: qdsp6: audioreach: simplify module_list sz calculation ASoC: qdsp6: audioreach: add support for more port connections ASoC: qdsp6: audioreach: add support to enable SAL Module ASoC: qdsp6: audioreach: add support for MFC Module ASoC: qdsp6: audioreach: add support to enable module command include/uapi/sound/snd_ar_tokens.h | 27 +++ sound/soc/qcom/qdsp6/audioreach.c | 310 ++++++++++++++++++++--------- sound/soc/qcom/qdsp6/audioreach.h | 47 +++-- sound/soc/qcom/qdsp6/q6apm.c | 84 +------- sound/soc/qcom/qdsp6/q6apm.h | 6 +- sound/soc/qcom/qdsp6/topology.c | 243 +++++++++++++++++++--- 6 files changed, 489 insertions(+), 228 deletions(-) -- 2.21.0

2 10

[PATCH 0/9] ASoC: qdsp6: audioreach: add multi-port, SAL and MFC support
by Srinivas Kandagatla 28 Oct '22

28 Oct '22

This patchset adds support to multi-port connections between AudioReach Modules which is required for sophisticated graphs like ECNS or Speaker Protection. Also as part of ECNS testing new module support for SAL and MFC are added. Tested on SM8450 with ECNS. Thanks, Srini Srinivas Kandagatla (9): ASoC: qdsp6: audioreach: topology use idr_alloc_u32 ASoC: qdsp6: audioreach: remove unused connection_list ASoC: qdsp6: audioreach: update dapm kcontrol private data ASoC: qdsp6: audioreach: Simplify handing FE and BE graph connections ASoC: qdsp6: audioreach: simplify module_list sz calculation ASoC: qdsp6: audioreach: add support for more port connections ASoC: qdsp6: audioreach: add support to enable SAL Module ASoC: qdsp6: audioreach: add support for MFC Module ASoC: qdsp6: audioreach: add support to enable module command include/uapi/sound/snd_ar_tokens.h | 27 +++ sound/soc/qcom/qdsp6/audioreach.c | 298 ++++++++++++++++++++--------- sound/soc/qcom/qdsp6/audioreach.h | 47 +++-- sound/soc/qcom/qdsp6/q6apm.c | 84 +------- sound/soc/qcom/qdsp6/q6apm.h | 6 +- sound/soc/qcom/qdsp6/topology.c | 242 +++++++++++++++++++---- 6 files changed, 480 insertions(+), 224 deletions(-) -- 2.21.0

3 11

[PATCH 00/21] ASoC: SOF: Intel: HDA: refactor codec and multi-link suport
by Pierre-Louis Bossart 28 Oct '22

28 Oct '22

Existing HDaudio controllers expose an HDAudio DMA which is used to interface with HDaudio codecs. All other interfaces supported by Intel (SoundWire, SSP, DMIC) rely for data transfers on another GP-DMA managed by the DSP firmware - the HDaudio DMA is only used for memory-to-DSP transfers. New HDaudio extensions will enable the use of this HDaudio DMA for all of SoundWire, SSP, DMIC. These extensions will be backwards-compatible for HDaudio and iDISP codecs, but will require new programming sequences and DAI callbacks for SoundWire, SSP and DMIC. Before we add support for 'extended audio links' and the programming sequences for the DMA, we need to refactor the code. All HDaudio codec support needs to be well identified in a separate file, and likewise all the 'multi-link' handling needs to be better split. This patchset removes a number of 'old' Kconfig dependencies and options, adds helpers with a fallback to remove IS_ENABLED checks in the code and tries to simplify programming sequences when possible. One indirect benefit from this refactoring is that developers can switch with a kernel parameter from HDaudio support to a variant of 'nocodec' support. This proves extremely useful to test on existing Intel RVPs and Up boards, where the same build can be used to check 3 interfaces (HDaudio, SSP, DMIC) by just removing modules, setting the kernel parameter and reloading modules. Pierre-Louis Bossart (21): ASoC: SOF: Intel: remove option to disable the common_hdmi handling ASoC: SOF: Intel: remove all dependencies on SND_SOC_HDAC_HDMI ASoC: SOF: Intel: hda-codec: simplify SND_SOC_SOF_HDA_AUDIO_CODEC handling ASoC: SOF: Intel: move codec state change to hda-codec.c ASoC: SOF: Intel: start moving multi-link handling in dedicated file ASoC: SOF: Intel: hda: add multi-link helper for LOSVID ASoC: SOF: Intel: move all RIRB/CMD_IO helpers to hda-codec.c ASoC: SOF: Intel: hda-ctrl: add codec wakeup helper ASoC: SOF: Intel: hda-codec: add hda_codec_device_remove() helper ASoC: SOF: Intel: hda-codec: add stop_cmd_io helper ASoC: SOF: Intel: hda-stream: always allocate CORB/RIRB buffer ASoC: SOF: Intel: hda-codec: add hda_codec_check_rirb_status() helper ASoC: SOF: Intel: hda-ctrl: use helper to clear RIRB status ASoC: SOF: Intel: hda-dsp: clarify dependencies on SND_SOC_SOF_HDA ASoC: SOF: Intel: hda-codec: add helpers to suspend and resume cmd_io ASoC: SOF: Intel: clarify bus_init and bus_exit sequences ASoC: SOF: Intel: hda-mlink: add helpers to suspend/resume links ASoC: SOF: Intel: add hda_bus_ml_free helper ASoC: SOF: Intel: hda: clarify Kconfig dependencies ASoC: SOF: Intel: hda-codec: use GPL-2.0-only license ASoC: SOF: introduce new DEBUG_NOCODEC mode sound/soc/sof/Kconfig | 18 ++- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/hda-bus.c | 23 ++- sound/soc/sof/intel/hda-codec.c | 252 +++++++++++++++++++++++++------ sound/soc/sof/intel/hda-ctrl.c | 63 ++------ sound/soc/sof/intel/hda-dai.c | 10 +- sound/soc/sof/intel/hda-dsp.c | 44 ++---- sound/soc/sof/intel/hda-mlink.c | 89 +++++++++++ sound/soc/sof/intel/hda-stream.c | 30 +--- sound/soc/sof/intel/hda.c | 62 ++------ sound/soc/sof/intel/hda.h | 63 ++++++-- sound/soc/sof/sof-audio.c | 5 + sound/soc/sof/sof-priv.h | 3 + 14 files changed, 438 insertions(+), 228 deletions(-) create mode 100644 sound/soc/sof/intel/hda-mlink.c -- 2.34.1

2 22

[PATCH] ASoC: fsl_asrc fsl_esai fsl_sai: allow CONFIG_PM=N
by Maarten Zanders 28 Oct '22

28 Oct '22

When CONFIG_PM=N, pm_runtime_put_sync() returns -ENOSYS which breaks the probe function of these drivers. Other users of pm_runtime_put_sync() typically don't check the return value. In order to keep the program flow as intended, check for -ENOSYS. This commit is similar to commit 0434d3f (omap-mailbox.c). This commit fixes: cab04ab (ASoC: fsl_asrc: Don't use devm_regmap_init_mmio_clk) 203773e (ASoC: fsl_esai: Don't use devm_regmap_init_mmio_clk) 2277e7e (ASoC: fsl_sai: Don't use devm_regmap_init_mmio_clk) Signed-off-by: Maarten Zanders <maarten.zanders(a)mind.be> --- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 44dcbf49456c..08ca410ef551 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1226,7 +1226,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; ret = devm_snd_soc_register_component(&pdev->dev, &fsl_asrc_component, diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bda66b30e063..763f5f0592af 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1070,7 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev) regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); ret = pm_runtime_put_sync(&pdev->dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /* diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 38f6362099d5..01f3463c0e47 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1149,7 +1149,7 @@ static int fsl_sai_probe(struct platform_device *pdev) } ret = pm_runtime_put_sync(dev); - if (ret < 0) + if (ret < 0 && ret != -ENOSYS) goto err_pm_get_sync; /*

3 6

[PATCH v1 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 28 Oct '22

28 Oct '22

v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 6 + .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 104 +++++++++++++++++- 2 files changed, 109 insertions(+), 1 deletion(-) -- 2.25.1

2 4

[PATCH v4 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 28 Oct '22

28 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v4: - replaced readb_poll_timeout() with read_poll_timeout() for the DRSM polling function as the register is u32 wide, not u8 Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 22 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 461 insertions(+), 60 deletions(-) -- 2.25.1

3 11

[PATCH] ASoC: fsl_xcvr: Add Counter registers
by Shengjiu Wang 28 Oct '22

28 Oct '22

These counter registers are part of register list, add them to complete the register map - DMAC counter control registers - Data path Timestamp counter register - Data path bit counter register - Data path bit count timestamp register - Data path bit read timestamp register Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 21 +++++++++++++++++++++ 2 files changed, 61 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c043efe4548d..2a6802fb2a8b 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -934,6 +934,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_RX_DPTH_CTRL_SET, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_CLR, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_TOG, 0x00002C89 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_RX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_SET, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_CLR, 0x00000000 }, @@ -944,6 +952,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_TX_CS_DATA_3, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_4, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_5, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_TX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_DEBUG_REG_0, 0x00000000 }, { FSL_XCVR_DEBUG_REG_1, 0x00000000 }, }; @@ -975,6 +991,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_RX_DPTH_TSCR: + case FSL_XCVR_RX_DPTH_BCR: + case FSL_XCVR_RX_DPTH_BCTR: + case FSL_XCVR_RX_DPTH_BCRR: case FSL_XCVR_TX_DPTH_CTRL: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: @@ -985,6 +1009,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_TX_DPTH_TSCR: + case FSL_XCVR_TX_DPTH_BCR: + case FSL_XCVR_TX_DPTH_BCTR: + case FSL_XCVR_TX_DPTH_BCRR: case FSL_XCVR_DEBUG_REG_0: case FSL_XCVR_DEBUG_REG_1: return true; @@ -1017,6 +1049,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: case FSL_XCVR_TX_DPTH_CTRL_TOG: @@ -1026,6 +1062,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: return true; default: return false; diff --git a/sound/soc/fsl/fsl_xcvr.h b/sound/soc/fsl/fsl_xcvr.h index 7f2853c60085..4769b0fca21d 100644 --- a/sound/soc/fsl/fsl_xcvr.h +++ b/sound/soc/fsl/fsl_xcvr.h @@ -49,6 +49,16 @@ #define FSL_XCVR_RX_DPTH_CTRL_CLR 0x188 #define FSL_XCVR_RX_DPTH_CTRL_TOG 0x18c +#define FSL_XCVR_RX_DPTH_CNTR_CTRL 0x1C0 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_SET 0x1C4 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR 0x1C8 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG 0x1CC + +#define FSL_XCVR_RX_DPTH_TSCR 0x1D0 +#define FSL_XCVR_RX_DPTH_BCR 0x1D4 +#define FSL_XCVR_RX_DPTH_BCTR 0x1D8 +#define FSL_XCVR_RX_DPTH_BCRR 0x1DC + #define FSL_XCVR_TX_DPTH_CTRL 0x220 /* TX datapath ctrl reg */ #define FSL_XCVR_TX_DPTH_CTRL_SET 0x224 #define FSL_XCVR_TX_DPTH_CTRL_CLR 0x228 @@ -59,6 +69,17 @@ #define FSL_XCVR_TX_CS_DATA_3 0x23C #define FSL_XCVR_TX_CS_DATA_4 0x240 #define FSL_XCVR_TX_CS_DATA_5 0x244 + +#define FSL_XCVR_TX_DPTH_CNTR_CTRL 0x260 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_SET 0x264 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR 0x268 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG 0x26C + +#define FSL_XCVR_TX_DPTH_TSCR 0x270 +#define FSL_XCVR_TX_DPTH_BCR 0x274 +#define FSL_XCVR_TX_DPTH_BCTR 0x278 +#define FSL_XCVR_TX_DPTH_BCRR 0x27C + #define FSL_XCVR_DEBUG_REG_0 0x2E0 #define FSL_XCVR_DEBUG_REG_1 0x2F0 -- 2.34.1

2 1

[PATCH 0/3] Add support for SAI on i.MX93 platform
by Chancel Liu 28 Oct '22

28 Oct '22

This patchset supports SAI on i.MX93 platform. Chancel Liu (3): ASoC: dt-bindings: fsl,sai: Add compatible string for i.MX93 platform ASoC: fsl_sai: Add support for i.MX93 platform ASoC: fsl_sai: Specify the maxburst to 8 on i.MX93 platform .../devicetree/bindings/sound/fsl,sai.yaml | 1 + sound/soc/fsl/fsl_sai.c | 23 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 21 insertions(+), 4 deletions(-) -- 2.25.1

5 8

[PATCH v1] ALSA: hda/realtek: Add quirk for ASUS Zenbook using CS35L41
by Stefan Binding 28 Oct '22

28 Oct '22

This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using I2C with Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701a72ec5629a..b4f7ff8cfe41b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9404,6 +9404,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.34.1

2 1

[PATCH 0/6] soundwire: qcom: fixes and support for v1.7
by Srinivas Kandagatla 28 Oct '22

28 Oct '22

This patchset has 2 fixes around fifo handling adds support for v1.7.0 controller found in SM8450 SoCs. Also a patch to make reset optional for v1.6 as this is not required for sc8280xp. Thanks, Srini Srinivas Kandagatla (6): soundwire: qcom: reinit broadcast completion soundwire: qcom: check for outanding writes before doing a read soundwire: qcom: remove unused SWRM_SPECIAL_CMD_ID soundwire: qcom: make reset optional for v1.6 controller dt-bindings: soundwire: qcom: add v1.7.0 support soundwire: qcom: add support for v1.7 Soundwire Controller .../bindings/soundwire/qcom,sdw.txt | 1 + drivers/soundwire/qcom.c | 34 ++++++++++++++++--- 2 files changed, 30 insertions(+), 5 deletions(-) -- 2.21.0

4 11

alsamixer not readable in xterm
by GitHub issues - opened 28 Oct '22

28 Oct '22

alsa-project/alsa-utils issue #176 was opened from Frank-Steiner: Hi, after upgrading from opensuse 15.3 to 15.4 we realized alsamixer is not useable in standard xterm with white background (which is our default) because the black background vanished. Found https://mailman.alsa-project.org/pipermail/alsa-devel/2021-June/185925.html which explains it. But that patch was a bad idea for everyone just using standard xterms and not themed ones. Adding sth. like a "--black-background" switch could be helpful for those people not working with themed X11... cu, Frank Issue URL : https://github.com/alsa-project/alsa-utils/issues/176 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[RFC][PATCH v2 29/31] timers: ALSA: Use del_timer_shutdown() before freeing timer
by Steven Rostedt 28 Oct '22

28 Oct '22

From: "Steven Rostedt (Google)" <rostedt(a)goodmis.org> Before a timer is freed, del_timer_shutdown() must be called. Link: https://lore.kernel.org/all/20220407161745.7d6754b3@gandalf.local.home/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Austin Kim <austin.kim(a)lge.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Steven Rostedt (Google) <rostedt(a)goodmis.org> --- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 1bc43e927d82..5269ab7321a4 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -47,7 +47,7 @@ static void reg_dump(struct ak4117 *ak4117) static void snd_ak4117_free(struct ak4117 *chip) { - del_timer_sync(&chip->timer); + del_timer_shutdown(&chip->timer); kfree(chip); } diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index a870759d179e..a43025f466bb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -129,7 +129,7 @@ int snd_emux_free(struct snd_emux *emu) if (! emu) return -EINVAL; - del_timer_sync(&emu->tlist); + del_timer_shutdown(&emu->tlist); snd_emux_proc_free(emu); snd_emux_delete_virmidi(emu); -- 2.35.1

2 1

[PATCH] ASoc: cs42xx8-i2c.c: add module device table for of
by Peter Bergin 28 Oct '22

28 Oct '22

When trying to connect the device with the driver through of it is not working. The of_device_id is defined in cs42xx8.c but is not correctly included in cs42xx8-i2c.c. Also add the matching table for of in the i2c file. Signed-off-by: Peter Bergin <peter(a)berginkonsult.se> --- sound/soc/codecs/cs42xx8-i2c.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index cb06a06d48b0..6e8ee28d01f8 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -37,6 +37,13 @@ static int cs42xx8_i2c_remove(struct i2c_client *i2c) return 0; } +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); + static struct i2c_device_id cs42xx8_i2c_id[] = { {"cs42448", (kernel_ulong_t)&cs42448_data}, {"cs42888", (kernel_ulong_t)&cs42888_data}, -- 2.34.1

4 7

[PATCH v2] ASoC: dt-bindings: Document audio OF graph dai-tdm-slot-num dai-tdm-slot-width props
by Marek Vasut 27 Oct '22

27 Oct '22

Document dai-tdm-slot-num and dai-tdm-slot-width props as those are parsed by simple graph card and may therefore appear in audio OF graph node. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: devicetree(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org To: linux-arm-kernel(a)lists.infradead.org --- V2: Drop the definition refs and just fill in type and description --- .../devicetree/bindings/sound/audio-graph-port.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index bc46a95ed8400..64654ceef2089 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -74,6 +74,12 @@ patternProperties: convert-sample-format: $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $ref: /schemas/types.yaml#/definitions/uint32 dai-tdm-slot-width-map: description: Mapping of sample widths to slot widths. For hardware that cannot support a fixed slot width or a slot width always -- 2.35.1

4 4

[PATCH 00/16] ASoC: SOF: Intel: HDaudio cleanups
by Pierre-Louis Bossart 27 Oct '22

27 Oct '22

This is the part1 of my HDaudio cleanups, before the addition of to-be-announced HDaudio extensions. The patchset includes more consistent use of read/write/update helpers, removal of useless waits, structure members and programming sequences, removal of confusing sharing of private_data between FE and BE. Additional patches are coming to split the controller, codec and multi-link management functionality in well-identified files. Pierre-Louis Bossart (16): ASoC: SOF: ops: fallback to mmio in helpers ASoC: SOF: Intel: use mmio fallback for all platforms ASoC: SOF: ops: add readb/writeb helpers ASoC: SOF: ops: add snd_sof_dsp_updateb() helper ASoC: SOF: Intel: hda-dsp: use SOF helpers for consistency ASoC: SOF: Intel: hda-dai: start removing the use of runtime->private_data in BE ASoC: SOF: Intel: hda-dai: use component_get_drvdata to find hdac_bus ASoC: SOF: Intel: hda-dai: remove useless members in hda_pipe_params ASoC: SOF: Intel: hda-ctrl: remove useless sleep ASoC: SOF: Intel: hda: always do a full reset ASoC: SOF: Intel: hda: remove useless check on GCTL ASoC: SOF: Intel: hda-stream: use SOF helpers for consistency ASoC: SOF: Intel: hda-stream: rename CL_SD_CTL registers as SD_CTL ASoC: SOF: Intel: hda: use SOF helper for consistency ASoC: SOF: Intel: hda-stream: use snd_sof_dsp_updateb() helper ASoC: SOF: Intel: hda-stream: use readb/writeb for stream registers sound/soc/sof/intel/bdw.c | 6 +--- sound/soc/sof/intel/byt.c | 12 ++----- sound/soc/sof/intel/hda-common-ops.c | 6 +--- sound/soc/sof/intel/hda-ctrl.c | 41 ++++++++-------------- sound/soc/sof/intel/hda-dai.c | 31 +++++++---------- sound/soc/sof/intel/hda-dsp.c | 12 ++++--- sound/soc/sof/intel/hda-loader-skl.c | 30 ++++++++-------- sound/soc/sof/intel/hda-loader.c | 4 +-- sound/soc/sof/intel/hda-stream.c | 47 ++++++++++++------------- sound/soc/sof/intel/hda.c | 5 ++- sound/soc/sof/intel/hda.h | 25 +++++++------- sound/soc/sof/intel/pci-tng.c | 6 +--- sound/soc/sof/ops.h | 51 ++++++++++++++++++++-------- sound/soc/sof/sof-priv.h | 4 +++ 14 files changed, 136 insertions(+), 144 deletions(-) -- 2.34.1

3 20

Add Focusrite Scarlett 2i2 gen2
by GitHub pull_request - opened 27 Oct '22

27 Oct '22

alsa-project/alsa-ucm-conf pull request #234 was opened from superseed: This is basically #203 but for the 2i2. I don't really know what I'm doing, but with these changes, I get two separate mono inputs in Pipewire, which is what I was looking for. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/234 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/234.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[BUG][v6.1-rc2] [SOF] max98373 i2c-MX98373:00: Runtime PM usage count underflow!
by Lukasz Majczak 27 Oct '22

27 Oct '22

Hi, I have noticed Runtime PM usage count underflow on my Chromebook (12th Gen Intel(R) Core(TM) i3-1215U) running vanilla v6.1-rc2: [ 7.558178] max98373 i2c-MX98373:00: Runtime PM usage count underflow! [ 7.558188] max98373 i2c-MX98373:01: Runtime PM usage count underflow! [ 7.558206] snd-soc-dummy snd-soc-dummy: Runtime PM usage count underflow! [ 7.558421] snd-soc-dummy snd-soc-dummy: Runtime PM usage count underflow! Link to the full dmesg: https://gist.github.com/semihalf-majczak-lukasz/d6a6d61d5937f762a524ed50fed… Best regards, Lukasz

2 1

[PATCH] ALSA: Use del_timer_sync() before freeing timer
by Steven Rostedt 27 Oct '22

27 Oct '22

From: "Steven Rostedt (Google)" <rostedt(a)goodmis.org> The current code for freeing the emux timer is extremely dangerous: CPU0 CPU1 ---- ---- snd_emux_timer_callback() snd_emux_free() spin_lock(&emu->voice_lock) del_timer(&emu->tlist); <-- returns immediately spin_unlock(&emu->voice_lock); [..] kfree(emu); spin_lock(&emu->voice_lock); [BOOM!] Instead just use del_timer_sync() which will wait for the timer to finish before continuing. No need to check if the timer is active or not when doing so. This doesn't fix the race of a possible re-arming of the timer, but at least it won't use the data that has just been freed. Cc: stable(a)vger.kernel.org Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Steven Rostedt (Google) <rostedt(a)goodmis.org> --- sound/synth/emux/emux.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 5ed8e36d2e04..a2ee78809cfb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -131,10 +131,7 @@ int snd_emux_free(struct snd_emux *emu) if (! emu) return -EINVAL; - spin_lock_irqsave(&emu->voice_lock, flags); - if (emu->timer_active) - del_timer(&emu->tlist); - spin_unlock_irqrestore(&emu->voice_lock, flags); + del_timer_sync(&emu->tlist); snd_emux_proc_free(emu); snd_emux_delete_virmidi(emu); -- 2.35.1

4 4

[PATCH 0/2] ASoC: Intel: sof_rt5682: quirk auto detection
by Brent Lu 27 Oct '22

27 Oct '22

A new feature to compose quirk value according to the result of amplifier and SoC type detection at runtime. Brent Lu (2): ASoC: Intel: add helpers to detect SoCs ASoC: Intel: sof_rt5682: quirk auto detection include/linux/platform_data/x86/soc.h | 24 +++ sound/soc/intel/boards/sof_rt5682.c | 173 +++++++----------- .../intel/common/soc-acpi-intel-adl-match.c | 12 +- .../intel/common/soc-acpi-intel-jsl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 6 +- 5 files changed, 101 insertions(+), 120 deletions(-) -- 2.25.1

4 7

Re: [PATCH 6/6] ASoC: Intel: boards: add missing dependency from realtek-common
by Mark Brown 27 Oct '22

27 Oct '22

On Mon, Oct 17, 2022 at 08:17:12AM +0200, Adam Borowski wrote: > These boards use code from sound/soc/sof/sof-audio.c > > Fixes: e1d5e13324020c4b405e63cae34560c7992bec2e > Signed-off-by: Adam Borowski <kilobyte(a)angband.pl> Please include human readable descriptions of things like commits and issues being discussed in e-mail in your mails, this makes them much easier for humans to read especially when they have no internet access. I do frequently catch up on my mail on flights or while otherwise travelling so this is even more pressing for me than just being about making things a bit easier to read. You've not incldued a cover letter for this and only copied me on some patches, are there any dependency issues?

1 0

[PATCH v3 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 27 Oct '22

27 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 21 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 460 insertions(+), 60 deletions(-) -- 2.25.1

3 14

[PATCH] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 27 Oct '22

27 Oct '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/intel/boards/sof_es8336.c | 58 ++++++++++++++++++++++------- 1 file changed, 44 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..7fde36a02d11 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,44 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) + gpiod_set_value_cansleep(priv->gpio_speakers, true); + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +161,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +371,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +751,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +781,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

2 1

[PATCH] ALSA: aoa: i2sbus: fix possible memory leak in i2sbus_add_dev()
by Yang Yingliang 27 Oct '22

27 Oct '22

dev_set_name() in soundbus_add_one() allocates memory for name, it need be freed when of_device_register() fails, call soundbus_dev_put() to give up the reference that hold in device_initialize(), so that it can be freed in kobject_cleanup() when the refcount hit to 0. And other resources are also freed in i2sbus_release_dev(), so it can return 0 directly. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index faf6b03131ee..f6841daf9e3b 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -302,6 +302,10 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (soundbus_add_one(&dev->sound)) { printk(KERN_DEBUG "i2sbus: device registration error!\n"); + if (dev->sound.ofdev.dev.kobj.state_initialized) { + soundbus_dev_put(&dev->sound); + return 0; + } goto err; } -- 2.25.1

2 3

[PATCH] ALSA: aoa: Fix I2S device accounting
by Takashi Iwai 27 Oct '22

27 Oct '22

i2sbus_add_dev() is supposed to return the number of probed devices, i.e. either 1 or 0. However, i2sbus_add_dev() has one error handling that returns -ENODEV; this will screw up the accumulation number counted in the caller, i2sbus_probe(). Fix the return value to 0 and add the comment for better understanding for readers. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/aoa/soundbus/i2sbus/core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index f6841daf9e3b..51ed2f34b276 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -147,6 +147,7 @@ static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index, return rc; } +/* Returns 1 if added, 0 for otherwise; don't return a negative value! */ /* FIXME: look at device node refcounting */ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_control *control, @@ -213,7 +214,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * either as the second one in that case is just a modem. */ if (!ok) { kfree(dev); - return -ENODEV; + return 0; } mutex_init(&dev->lock); -- 2.35.3

1 0

[GIT PULL] ASoC fixes for v6.1-rc2
by Mark Brown 27 Oct '22

27 Oct '22

The following changes since commit 9abf2313adc1ca1b6180c508c25f22f9395cc780: Linux 6.1-rc1 (2022-10-16 15:36:24 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.1-rc2 for you to fetch changes up to e9441675edc1bb8dbfadacf68aafacca60d65a25: ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() (2022-10-21 13:04:14 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.1 Quite a few fixes here, a lot driver specific, plus some new quirks. There was a bit of a mess with the runtime PM handling due to some confusion in the API there which resulted in a number of commits and reverts but that should all be stable now. ---------------------------------------------------------------- Aidan MacDonald (1): ASoC: simple-card: Fix up checks for HW param fixups Colin Ian King (1): ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source", "Routee" -> "Route" Derek Fang (2): ASoC: rt5682s: Fix the TDM Tx settings ASoC: rt1019: Fix the TDM settings Geert Uytterhoeven (1): ASoC: codecs: tlv320adc3xxx: Wrap adc3xxx_i2c_remove() in __exit_p() Jiangshan Yi (1): ASoC: cx2072x: fix spelling typo in comment Kai Vehmanen (1): ASoC: SOF: ipc4-mtrace: protect per-core nodes against multiple open Leohearts (1): ASoC: amd: yc: Add Lenovo Thinkbook 14+ 2022 21D0 to quirks table Mark Brown (2): ASoC: wmxxxx: Revert old "ASoC: wmxxxx: Fix PM disable depth imbalance in wmxxxx_probe" ASoC: jz4752b: Capture fixes Peter Ujfalusi (1): Revert "ASoC: soc-component: using pm_runtime_resume_and_get instead of pm_runtime_get_sync" Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: add quirk variant for LAPBC710 NUC15 ASoC: SOF: Intel: pci-mtl: fix firmware name ASoC: SOF: Intel: pci-tgl: fix ADL-N descriptor Randy Dunlap (2): ASoC: codec: tlv320adc3xxx: add GPIOLIB dependency ASoC: qcom: SND_SOC_SC7180 optionally depends on SOUNDWIRE Shuming Fan (2): ASoC: rt1308-sdw: update the preset settings ASoC: rt1308-sdw: add the default value of some registers Siarhei Volkau (4): ASoC: codecs: jz4725b: add missed Line In power control bit ASoC: codecs: jz4725b: fix reported volume for Master ctl ASoC: codecs: jz4725b: use right control for Capture Volume ASoC: codecs: jz4725b: fix capture selector naming Srinivasa Rao Mandadapu (2): ASoC: qcom: lpass-cpu: mark HDMI TX registers as volatile ASoC: qcom: lpass-cpu: Mark HDMI TX parity register as volatile Xiaolei Wang (1): ASoC: wm8962: Add an event handler for TEMP_HP and TEMP_SPK Yang Yingliang (2): ASoC: Intel: Skylake: fix possible memory leak in skl_codec_device_init() ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() Yong Zhi (1): ASoC: Intel: sof_rt5682: Add quirk for Rex board Zhang Qilong (7): ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe" ASoC: wm5110: Revert "ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe" ASoC: wm8997: Revert "ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe" ASoC: mt6660: Keep the pm_runtime enables before component stuff in mt6660_i2c_probe linkt (1): ASoC: amd: yc: Adding Lenovo ThinkBook 14 Gen 4+ ARA and Lenovo ThinkBook 16 Gen 4+ ARA to the Quirks List include/sound/simple_card_utils.h | 1 + sound/soc/amd/yc/acp6x-mach.c | 21 ++++++++++++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cx2072x.h | 2 +- sound/soc/codecs/jz4725b.c | 34 ++++++++++++---------- sound/soc/codecs/mt6660.c | 8 +++--- sound/soc/codecs/rt1019.c | 20 +++++++------ sound/soc/codecs/rt1019.h | 6 ++++ sound/soc/codecs/rt1308-sdw.c | 17 +++++++++-- sound/soc/codecs/rt1308-sdw.h | 3 ++ sound/soc/codecs/rt1308.h | 5 ++++ sound/soc/codecs/rt5682s.c | 15 ++++++++-- sound/soc/codecs/rt5682s.h | 1 + sound/soc/codecs/tlv320adc3xxx.c | 2 +- sound/soc/codecs/wm5102.c | 7 +++-- sound/soc/codecs/wm5110.c | 7 +++-- sound/soc/codecs/wm8962.c | 54 +++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8997.c | 7 +++-- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 15 ++++++++++ sound/soc/generic/simple-card.c | 3 +- sound/soc/intel/boards/sof_rt5682.c | 12 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++ sound/soc/intel/skylake/skl.c | 8 +----- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/lpass-cpu.c | 10 +++++++ sound/soc/soc-component.c | 6 ++-- sound/soc/sof/intel/hda-codec.c | 8 +----- sound/soc/sof/intel/pci-mtl.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 30 ++++++++++++++++++- sound/soc/sof/ipc4-mtrace.c | 20 +++++++++++-- 31 files changed, 270 insertions(+), 69 deletions(-)

2 1

[Regression] Bug 216613 - Sound stopped working with v6.0.3 on Lenovo T14 Gen2i: ASoC: error at snd_soc_component_probe
by Thorsten Leemhuis 27 Oct '22

27 Oct '22

Hi, this is your Linux kernel regression tracker speaking. I noticed a regression report in bugzilla.kernel.org. As many (most?) kernel developer don't keep an eye on it, I decided to forward it by mail. Quoting from https://bugzilla.kernel.org/show_bug.cgi?id=216613 : > Grzegorz Alibożek 2022-10-21 19:26:43 UTC > > After upgrade kernel from 6.0.2 to 6.0.3 on Lenovo T14 Gen2i, sound stopped working. > dmesg: > > paź 21 21:11:45 kernel: snd_hda_codec_hdmi ehdaudio0D2: failed to create hda codec -12 > paź 21 21:11:45 kernel: snd_hda_codec_hdmi ehdaudio0D2: ASoC: error at snd_soc_component_probe on ehdaudio0D2: -12 > paź 21 21:11:45 kernel: skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: failed to instantiate card -12 > > [reply] [−] Comment 1 Grzegorz Alibożek 2022-10-21 19:56:43 UTC > > Created attachment 303070 [details] > trace See the ticket for more details. BTW, let me use this mail to also add the report to the list of tracked regressions to ensure it's doesn't fall through the cracks: #regzbot introduced: v6.0.2..v6.0.3 https://bugzilla.kernel.org/show_bug.cgi?id=216613 #regzbot ignore-activity Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) P.S.: As the Linux kernel's regression tracker I deal with a lot of reports and sometimes miss something important when writing mails like this. If that's the case here, don't hesitate to tell me in a public reply, it's in everyone's interest to set the public record straight.

2 3

Thanks to review question [PATCH V2 1/2] ASoc:codes:Add Awinic AW883XX audio amplifier driver
by wangweidong.a＠awinic.com 27 Oct '22

27 Oct '22

Hi Mark Brown Thank you for your valuable advice to let me understand the problem. I will fix this issue in the next patch Best regards Weidong Wang On Mon, Oct 24, 2022 at 10:41:03AM +0800, wangweidong.a(a)awinic.com wrote: > > Then it's not a mute function, the goal of the mute function is to > > run > before all the power management code to minimise glitches during power > management. Just implement the power management via the standard ASoC > power > > > management APIs. > The essence of calling mute is not switch dsp, but switch PA. We think > that PA has only two states. When no audio stream enters, turn off the > PA and turn off the PA's dsp. When the audio stream enters, the PA is > turned on, and the dsp is turned on at the same time > That's not what the mute function is for, the interface is specifically for muting the stream while power management changes are > going on. What you're describing is power management so should be controlled via DAPM. > If your device doesn't have any support for a separate mute function then it should just not implement anything for that in the > driver.

1 0

[PATCH v2 0/9] ASoC: qdsp6: audioreach: add multi-port, SAL and MFC support
by Srinivas Kandagatla 26 Oct '22

26 Oct '22

This patchset adds support to multi-port connections between AudioReach Modules which is required for sophisticated graphs like ECNS or Speaker Protection. Also as part of ECNS testing new module support for SAL and MFC are added. Tested on SM8450 with ECNS. Thanks, Srini Changes since v1: Fixed two warnings on unused and unintialized variable. Srinivas Kandagatla (9): ASoC: qdsp6: audioreach: topology use idr_alloc_u32 ASoC: qdsp6: audioreach: remove unused connection_list ASoC: qdsp6: audioreach: update dapm kcontrol private data ASoC: qdsp6: audioreach: Simplify handing FE and BE graph connections ASoC: qdsp6: audioreach: simplify module_list sz calculation ASoC: qdsp6: audioreach: add support for more port connections ASoC: qdsp6: audioreach: add support to enable SAL Module ASoC: qdsp6: audioreach: add support for MFC Module ASoC: qdsp6: audioreach: add support to enable module command include/uapi/sound/snd_ar_tokens.h | 27 +++ sound/soc/qcom/qdsp6/audioreach.c | 310 ++++++++++++++++++++--------- sound/soc/qcom/qdsp6/audioreach.h | 47 +++-- sound/soc/qcom/qdsp6/q6apm.c | 84 +------- sound/soc/qcom/qdsp6/q6apm.h | 6 +- sound/soc/qcom/qdsp6/topology.c | 243 +++++++++++++++++++--- 6 files changed, 489 insertions(+), 228 deletions(-) -- 2.21.0

2 13

[PATCH v1 1/2] ASoC: simple-card: Support custom DAI system clock IDs
by Aidan MacDonald 26 Oct '22

26 Oct '22

Some DAIs have multiple system clock sources, which can be chosen using the "clk_id" argument to snd_soc_dai_set_sysclk(). Currently this is hardcoded to 0 when using simple cards, but that choice is not always suitable. Add the "system-clock-id" property to allow selecting a different clock ID on a per-DAI basis. To simplify the logic on DPCM cards, add a dummy "asoc_simple_dai" instance and use that for the dummy components on DPCM links. This ensures that when we're iterating over DAIs in the PCM runtime there is always a matching "asoc_simple_dai" we can dereference. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 26 ++++++++++++++++++++------ 2 files changed, 22 insertions(+), 6 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a0b827f0c2f6..9f9a72299637 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -26,6 +26,7 @@ struct asoc_simple_dai { const char *name; unsigned int sysclk; int clk_direction; + int sysclk_id; int slots; int slot_width; unsigned int tx_slot_mask; @@ -67,6 +68,7 @@ struct asoc_simple_priv { struct prop_nums num; unsigned int mclk_fs; } *dai_props; + struct asoc_simple_dai dummy_dai; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bef16833c487..d4d898e06e76 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -262,6 +262,9 @@ int asoc_simple_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; + if (!of_property_read_u32(node, "system-clock-id", &val)) + simple_dai->sysclk_id = val; + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_parse_clk); @@ -355,7 +358,7 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(cpu_dai)) snd_soc_dai_set_sysclk(cpu_dai, - 0, 0, SND_SOC_CLOCK_OUT); + dai->sysclk_id, 0, SND_SOC_CLOCK_OUT); asoc_simple_clk_disable(dai); } @@ -364,7 +367,7 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(codec_dai)) snd_soc_dai_set_sysclk(codec_dai, - 0, 0, SND_SOC_CLOCK_IN); + dai->sysclk_id, 0, SND_SOC_CLOCK_IN); asoc_simple_clk_disable(dai); } @@ -439,7 +442,7 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; - int i, ret; + int i, ret, sysclk_id; if (props->mclk_fs) mclk_fs = props->mclk_fs; @@ -472,13 +475,21 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, } for_each_rtd_codec_dais(rtd, i, sdai) { - ret = snd_soc_dai_set_sysclk(sdai, 0, mclk, SND_SOC_CLOCK_IN); + pdai = simple_props_to_dai_codec(props, i); + sysclk_id = pdai->sysclk_id; + + ret = snd_soc_dai_set_sysclk(sdai, sysclk_id, mclk, + SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) return ret; } for_each_rtd_cpu_dais(rtd, i, sdai) { - ret = snd_soc_dai_set_sysclk(sdai, 0, mclk, SND_SOC_CLOCK_OUT); + pdai = simple_props_to_dai_cpu(props, i); + sysclk_id = pdai->sysclk_id; + + ret = snd_soc_dai_set_sysclk(sdai, pdai->sysclk_id, mclk, + SND_SOC_CLOCK_OUT); if (ret && ret != -ENOTSUPP) return ret; } @@ -523,7 +534,8 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; if (simple_dai->sysclk) { - ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, + ret = snd_soc_dai_set_sysclk(dai, simple_dai->sysclk_id, + simple_dai->sysclk, simple_dai->clk_direction); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_sysclk error\n"); @@ -858,6 +870,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dai_link[i].cpus = &priv->dummy; dai_props[i].num.cpus = dai_link[i].num_cpus = 1; + dai_props[i].cpu_dai = &priv->dummy_dai; } if (li->num[i].codecs) { @@ -882,6 +895,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dai_link[i].codecs = &priv->dummy; dai_props[i].num.codecs = dai_link[i].num_codecs = 1; + dai_props[i].codec_dai = &priv->dummy_dai; } if (li->num[i].platforms) { -- 2.38.1

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[PATCH v6 0/9] ASoC: cleanups and improvements for jz4740-i2s
by Aidan MacDonald 26 Oct '22

26 Oct '22

This series is a preparatory cleanup of the jz4740-i2s driver before adding support for a new SoC. The two improvements are lifting unnecessary restrictions on sample rates and formats -- the existing ones appear to be derived from the limitations of the JZ4740's internal codec and don't reflect the actual capabilities of the I2S controller. I'm unable to test the series on any JZ47xx SoCs, but I have tested on an X1000 (which is the SoC I'll be adding in a followup series). Changes in v6: * Added Cc: stable for flush bits fix. Changes in v5: * Drop 'mem' resource removal patch already upstream. * Update FIFO flush bits fix to address Paul's review comments. * Drop PLL clock name patch, that needs a different approach. Link for v4: https://lore.kernel.org/alsa-devel/20220708160244.21933-1-aidanmacdonald.0x… Aidan MacDonald (9): ASoC: jz4740-i2s: Handle independent FIFO flush bits ASoC: jz4740-i2s: Convert to regmap API ASoC: jz4740-i2s: Simplify using regmap fields ASoC: jz4740-i2s: Use FIELD_PREP() macros in hw_params callback ASoC: jz4740-i2s: Align macro values and sort includes ASoC: jz4740-i2s: Support S20_LE and S24_LE sample formats ASoC: jz4740-i2s: Support continuous sample rate ASoC: jz4740-i2s: Move component functions near the component driver ASoC: jz4740-i2s: Refactor DAI probe/remove ops as component ops sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 455 ++++++++++++++++++---------------- 2 files changed, 243 insertions(+), 213 deletions(-) -- 2.38.1

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[PATCH] ASoC: dt-bindings: mt8192-mt6359: Set maxItems, not type, for sound-dai
by Nícolas F. R. A. Prado 26 Oct '22

26 Oct '22

sound-dai is a standard property whose type is already set to phandle-array by sound-dai.yaml, so there's no need to set it (and wrongly so for headset-codec) in this binding. What should be set however is the maximum number of items, which for headset-codec should be 1. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- .../devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 478be7e3fa29..c6e614c1c30b 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -34,7 +34,7 @@ properties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 required: - sound-dai @@ -48,7 +48,6 @@ properties: maxItems: 2 items: maxItems: 1 - $ref: /schemas/types.yaml#/definitions/phandle-array required: - sound-dai -- 2.38.1

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[PATCH] ASoC: SOF: ipc4-loader: Return ssize_t from sof_ipc4_fw_parse_ext_man()
by Peter Ujfalusi 26 Oct '22

26 Oct '22

sof_ipc4_fw_parse_ext_man() can return negative error numbers which is not correct for the used size_t type. Change the return value to ssize_t and use the same type where the function is called. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 73c091a2fe96 ("ASoC: SOF: ipc4-loader: Support for loading external libraries") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index af0018b38cf0..702196774c50 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -17,8 +17,8 @@ /* The module ID includes the id of the library it is part of at offset 12 */ #define SOF_IPC4_MOD_LIB_ID_SHIFT 12 -static size_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev, - struct sof_ipc4_fw_library *fw_lib) +static ssize_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev, + struct sof_ipc4_fw_library *fw_lib) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; const struct firmware *fw = fw_lib->sof_fw.fw; @@ -141,7 +141,7 @@ static size_t sof_ipc4_fw_parse_basefw_ext_man(struct snd_sof_dev *sdev) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct sof_ipc4_fw_library *fw_lib; - size_t payload_offset; + ssize_t payload_offset; int ret; fw_lib = devm_kzalloc(sdev->dev, sizeof(*fw_lib), GFP_KERNEL); @@ -170,7 +170,7 @@ static int sof_ipc4_load_library_by_uuid(struct snd_sof_dev *sdev, struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct sof_ipc4_fw_library *fw_lib; const char *fw_filename; - size_t payload_offset; + ssize_t payload_offset; int ret, i, err; if (!sdev->pdata->fw_lib_prefix) { -- 2.38.1

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[PATCH] ASoC: SOF: Intel: MTL: fix comment error
by Pierre-Louis Bossart 26 Oct '22

26 Oct '22

From: Yong Zhi <yong.zhi(a)intel.com> Fix small cut-and-paste error in comment. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/mtl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 7e8b298786df6..91619036762b4 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -393,7 +393,7 @@ static int mtl_dsp_core_power_down(struct snd_sof_dev *sdev, int core) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK, 0); - /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + /* Wait for unstable CPA read (0 then 1 then 0) just after setting SPA bit */ usleep_range(1000, 1010); ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, dspcxctl, @@ -422,7 +422,7 @@ static int mtl_power_down_dsp(struct snd_sof_dev *sdev) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFDSSCS, MTL_HFDSSCS_SPA_MASK, 0); - /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + /* Wait for unstable CPA read (0 then 1 then 0) just after setting SPA bit */ usleep_range(1000, 1010); /* poll with timeout to check if operation successful */ -- 2.34.1

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[PATCH 0/2] ASoC: Intel: Fix typo in comments
by Amadeusz Sławiński 26 Oct '22

26 Oct '22

There is a typo which keeps propagating through existing Intel machine boards fix it with hope that fix stops it from appearing in new boards. Amadeusz Sławiński (2): ASoC: Intel: boards: Fix typo in comments ASoC: Intel: avs: boards: Fix typo in comments sound/soc/intel/avs/boards/max98373.c | 2 +- sound/soc/intel/avs/boards/max98927.c | 2 +- sound/soc/intel/avs/boards/ssm4567.c | 2 +- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- 17 files changed, 17 insertions(+), 17 deletions(-) -- 2.25.1

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[PATCH] ASoC: amd: yc: Add Alienware m17 R5 AMD into DMI table
by Brent Mendelsohn 26 Oct '22

26 Oct '22

This model requires an additional detection quirk to enable the internal microphone - BIOS doesn't seem to support AcpDmicConnected (nothing in acpidump output). Link: https://bugzilla.kernel.org/show_bug.cgi?id=216590 Signed-off-by: Brent Mendelsohn <mendiebm(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 2cb50d5cf1a9..a868a890f00c 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -185,6 +185,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Alienware"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alienware m17 R5 AMD"), + } + }, {} }; -- 2.37.3

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[PATCH] ASoC: Intel: bytcht_es8316: Add quirk for the Nanote UMPC-01
by Hans de Goede 26 Oct '22

26 Oct '22

The Nanote UMPC-01 mini laptop has stereo speakers, while the default bytcht_es8316 settings assume a mono speaker setup. Add a quirk for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcht_es8316.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 6432b83f616f..a935c5fd9edb 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -443,6 +443,13 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_JD_INVERTED), }, + { /* Nanote UMPC-01 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "RWC CO.,LTD"), + DMI_MATCH(DMI_PRODUCT_NAME, "UMPC-01"), + }, + .driver_data = (void *)BYT_CHT_ES8316_INTMIC_IN1_MAP, + }, { /* Teclast X98 Plus II */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), -- 2.37.3

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