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[PATCH AUTOSEL 5.18 39/68] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Sasha Levin 07 Jun '22

07 Jun '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 832296804bc7171730884e78c761c29f6d258e13 ] The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20220516103055.20003-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 30c2e7cb7ed2..3559d9ecfa07 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2094,12 +2094,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2108,9 +2110,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2535,6 +2539,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d76a505052fb..96a2ea92dd1e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1179,12 +1179,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.1

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[PATCH AUTOSEL 5.18 37/68] ASoC: SOF: amd: Fixed Build error
by Sasha Levin 07 Jun '22

07 Jun '22

From: Syed Saba kareem <ssabakar(a)amd.com> [ Upstream commit 803a1f7272797faa15a7879cdc70f9adaf3fdcba ] Add linux/module.h in acp-pci.c to solve the below dependency All error/warnings (new ones prefixed by >>): >> sound/soc/amd/acp/acp-pci.c:148:1: warning: data definition has no type or storage class 148 | MODULE_DEVICE_TABLE(pci, acp_pci_ids); | ^~~~~~~~~~~~~~~~~~~ >> sound/soc/amd/acp/acp-pci.c:148:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int] ... Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Syed Saba Kareem<ssabakar(a)amd.com> Link: https://lore.kernel.org/r/20220523112956.3087604-1-ssabakar@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 340e39d7f420..c893963ee2d0 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -16,6 +16,7 @@ #include <linux/pci.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> +#include <linux/module.h> #include "amd.h" #include "../mach-config.h" -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 22/68] soundwire: qcom: adjust autoenumeration timeout
by Sasha Levin 07 Jun '22

07 Jun '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 74da272400b46f2e898f115d1b1cd60828766919 ] Currently timeout for autoenumeration during probe and bus reset is set to 2 secs which is really a big value. This can have an adverse effect on boot time if the slave device is not ready/reset. This was the case with wcd938x which was not reset yet but we spent 2 secs waiting in the soundwire controller probe. Reduce this time to 1/10 of Hz which should be good enough time to finish autoenumeration if any slaves are available on the bus. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20220506084705.18525-1-srinivas.kandagatla@linaro… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7ebb1aa..22947e8297ac 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -105,7 +105,7 @@ #define SWRM_SPECIAL_CMD_ID 0xF #define MAX_FREQ_NUM 1 -#define TIMEOUT_MS (2 * HZ) +#define TIMEOUT_MS 100 #define QCOM_SWRM_MAX_RD_LEN 0x1 #define QCOM_SDW_MAX_PORTS 14 #define DEFAULT_CLK_FREQ 9600000 -- 2.35.1

1 0

[PATCH v2] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/G2UL SoC
by Biju Das 07 Jun '22

07 Jun '22

Document RZ/G2U2L SSI bindings. RZ/G2UL SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. While at it add a '.' at the end of dmas description for the first cell. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Acked-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Reviewed-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v1->v2: * Rebased to latest sound.git * Added Ack tag from Krzysztof Kozlowski * Added Rb tag from Geert. --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 7e8d252f7bca..0d9840375132 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -13,6 +13,7 @@ properties: compatible: items: - enum: + - renesas,r9a07g043-ssi # RZ/G2UL - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi @@ -50,7 +51,7 @@ properties: minItems: 1 maxItems: 2 description: - The first cell represents a phandle to dmac + The first cell represents a phandle to dmac. The second cell specifies the encoded MID/RID values of the SSI port connected to the DMA client and the slave channel configuration parameters. -- 2.25.1

2 1

[syzbot] KASAN: use-after-free Write in snd_rawmidi_kernel_write1
by syzbot 07 Jun '22

07 Jun '22

Hello, syzbot found the following issue on: HEAD commit: 952923ddc011 Merge tag 'pull-18-rc1-work.namei' of git://g.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=158076dbf00000 kernel config: https://syzkaller.appspot.com/x/.config?x=3096247591885bfa dashboard link: https://syzkaller.appspot.com/bug?extid=8569b906285a4215278d compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+8569b906285a4215278d(a)syzkaller.appspotmail.com ================================================================== ================================================================== BUG: KASAN: use-after-free in instrument_copy_from_user include/linux/instrumented.h:135 [inline] BUG: KASAN: use-after-free in _copy_from_user+0x12a/0x170 lib/usercopy.c:15 Write of size 4096 at addr ffff8880280e2000 by task syz-executor.2/8284 CPU: 1 PID: 8284 Comm: syz-executor.2 Not tainted 5.18.0-syzkaller-13842-g952923ddc011 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_address_description.constprop.0.cold+0xeb/0x495 mm/kasan/report.c:313 print_report mm/kasan/report.c:429 [inline] kasan_report.cold+0xf4/0x1c6 mm/kasan/report.c:491 check_region_inline mm/kasan/generic.c:183 [inline] kasan_check_range+0x13d/0x180 mm/kasan/generic.c:189 instrument_copy_from_user include/linux/instrumented.h:135 [inline] _copy_from_user+0x12a/0x170 lib/usercopy.c:15 copy_from_user include/linux/uaccess.h:152 [inline] snd_rawmidi_kernel_write1+0x361/0x870 sound/core/rawmidi.c:1486 snd_rawmidi_write+0x273/0xb90 sound/core/rawmidi.c:1555 vfs_write+0x269/0xac0 fs/read_write.c:589 ksys_write+0x1e8/0x250 fs/read_write.c:644 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+0x46/0xb0 RIP: 0033:0x7fe46fa89109 Code: ff ff c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 40 00 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007fe470c0a168 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007fe46fb9bf60 RCX: 00007fe46fa89109 RDX: 00000000fffffd2c RSI: 0000000020000000 RDI: 0000000000000004 RBP: 00007fe46fae308d R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 R13: 00007fe4700cfb1f R14: 00007fe470c0a300 R15: 0000000000022000 </TASK> Allocated by task 8284: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:45 [inline] set_alloc_info mm/kasan/common.c:436 [inline] ____kasan_kmalloc mm/kasan/common.c:515 [inline] ____kasan_kmalloc mm/kasan/common.c:474 [inline] __kasan_kmalloc+0xa9/0xd0 mm/kasan/common.c:524 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x3e/0x190 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] kvzalloc include/linux/slab.h:758 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:162 [inline] open_substream+0x340/0x8b0 sound/core/rawmidi.c:306 rawmidi_open_priv+0x591/0x6f0 sound/core/rawmidi.c:357 snd_rawmidi_open+0x49f/0xb60 sound/core/rawmidi.c:455 snd_open+0x21f/0x460 sound/core/sound.c:169 chrdev_open+0x266/0x770 fs/char_dev.c:414 do_dentry_open+0x4a1/0x11f0 fs/open.c:848 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 do_sys_openat2+0x16d/0x4c0 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1305 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+0x46/0xb0 Freed by task 8283: kasan_save_stack+0x1e/0x40 mm/kasan/common.c:38 kasan_set_track+0x21/0x30 mm/kasan/common.c:45 kasan_set_free_info+0x20/0x30 mm/kasan/generic.c:370 ____kasan_slab_free mm/kasan/common.c:366 [inline] ____kasan_slab_free+0x166/0x1a0 mm/kasan/common.c:328 kasan_slab_free include/linux/kasan.h:200 [inline] slab_free_hook mm/slub.c:1727 [inline] slab_free_freelist_hook+0x8b/0x1c0 mm/slub.c:1753 slab_free mm/slub.c:3507 [inline] kfree+0xd6/0x4d0 mm/slub.c:4555 kvfree+0x42/0x50 mm/util.c:655 snd_rawmidi_runtime_free sound/core/rawmidi.c:176 [inline] close_substream.part.0+0x185/0x720 sound/core/rawmidi.c:528 close_substream sound/core/rawmidi.c:507 [inline] rawmidi_release_priv+0x192/0x270 sound/core/rawmidi.c:547 snd_rawmidi_release+0x5e/0xf0 sound/core/rawmidi.c:578 __fput+0x277/0x9d0 fs/file_table.c:317 task_work_run+0xdd/0x1a0 kernel/task_work.c:177 resume_user_mode_work include/linux/resume_user_mode.h:49 [inline] exit_to_user_mode_loop kernel/entry/common.c:169 [inline] exit_to_user_mode_prepare+0x23c/0x250 kernel/entry/common.c:201 __syscall_exit_to_user_mode_work kernel/entry/common.c:283 [inline] syscall_exit_to_user_mode+0x19/0x50 kernel/entry/common.c:294 do_syscall_64+0x42/0xb0 arch/x86/entry/common.c:86 entry_SYSCALL_64_after_hwframe+0x46/0xb0 The buggy address belongs to the object at ffff8880280e2000 which belongs to the cache kmalloc-4k of size 4096 The buggy address is located 0 bytes inside of 4096-byte region [ffff8880280e2000, ffff8880280e3000) The buggy address belongs to the physical page: page:ffffea0000a03800 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x280e0 head:ffffea0000a03800 order:3 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000010200 dead000000000100 dead000000000122 ffff888011842140 raw: 0000000000000000 0000000000040004 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 3, migratetype Unmovable, gfp_mask 0x1d2040(__GFP_IO|__GFP_NOWARN|__GFP_NORETRY|__GFP_COMP|__GFP_NOMEMALLOC|__GFP_HARDWALL), pid 3912, tgid 3912 (udevd), ts 388242276639, free_ts 388232231340 prep_new_page mm/page_alloc.c:2456 [inline] get_page_from_freelist+0x1290/0x3b70 mm/page_alloc.c:4198 __alloc_pages+0x1c7/0x510 mm/page_alloc.c:5426 alloc_pages+0x1aa/0x310 mm/mempolicy.c:2272 alloc_slab_page mm/slub.c:1797 [inline] allocate_slab+0x26c/0x3c0 mm/slub.c:1942 new_slab mm/slub.c:2002 [inline] ___slab_alloc+0x985/0xd90 mm/slub.c:3002 __slab_alloc.constprop.0+0x4d/0xa0 mm/slub.c:3089 slab_alloc_node mm/slub.c:3180 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x318/0x350 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] tomoyo_realpath_from_path+0xc3/0x620 security/tomoyo/realpath.c:254 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x272/0x380 security/tomoyo/file.c:771 tomoyo_file_open security/tomoyo/tomoyo.c:320 [inline] tomoyo_file_open+0x9d/0xc0 security/tomoyo/tomoyo.c:315 security_file_open+0x45/0xb0 security/security.c:1645 do_dentry_open+0x349/0x11f0 fs/open.c:835 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 do_sys_openat2+0x16d/0x4c0 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x13f/0x1f0 fs/open.c:1305 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1371 [inline] free_pcp_prepare+0x549/0xd20 mm/page_alloc.c:1421 free_unref_page_prepare mm/page_alloc.c:3343 [inline] free_unref_page+0x19/0x6a0 mm/page_alloc.c:3438 __unfreeze_partials+0x17c/0x1a0 mm/slub.c:2521 qlink_free mm/kasan/quarantine.c:168 [inline] qlist_free_all+0x6a/0x170 mm/kasan/quarantine.c:187 kasan_quarantine_reduce+0x180/0x200 mm/kasan/quarantine.c:294 __kasan_slab_alloc+0xa2/0xc0 mm/kasan/common.c:446 kasan_slab_alloc include/linux/kasan.h:224 [inline] slab_post_alloc_hook mm/slab.h:750 [inline] slab_alloc_node mm/slub.c:3214 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x200/0x350 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] kzalloc include/linux/slab.h:733 [inline] tomoyo_encode2.part.0+0xe9/0x3a0 security/tomoyo/realpath.c:45 tomoyo_encode2 security/tomoyo/realpath.c:31 [inline] tomoyo_encode+0x28/0x50 security/tomoyo/realpath.c:80 tomoyo_realpath_from_path+0x186/0x620 security/tomoyo/realpath.c:288 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x272/0x380 security/tomoyo/file.c:771 tomoyo_file_open security/tomoyo/tomoyo.c:320 [inline] tomoyo_file_open+0x9d/0xc0 security/tomoyo/tomoyo.c:315 security_file_open+0x45/0xb0 security/security.c:1645 do_dentry_open+0x349/0x11f0 fs/open.c:835 do_open fs/namei.c:3520 [inline] path_openat+0x1c71/0x2910 fs/namei.c:3653 do_filp_open+0x1aa/0x400 fs/namei.c:3680 Memory state around the buggy address: ffff8880280e1f00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8880280e1f80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff8880280e2000: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8880280e2080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8880280e2100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- 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.

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[PATCH 0/3] ASoC: ssm: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

Trivial updates to modernise the various ssm* drivers to the new DAI clocking constants. Mark Brown (3): ASoC: ssm2518: Use modern ASoC DAI format terminology ASoC: ssm2602: Use modern ASoC DAI format terminology ASoC: ssm4567: Use modern ASoC DAI format terminology sound/soc/codecs/ssm2518.c | 4 ++-- sound/soc/codecs/ssm2602.c | 6 +++--- sound/soc/codecs/ssm4567.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) -- 2.30.2

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[PATCH 0/5] ASoC: SOF: AMD/Mediatek updates for 5.20
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

Small patches reviewed on SOF GitHub. Ajit Kumar Pandey (1): ASoC: SOF: amd: Add SOF pm ops callback for Renoir YC Hung (4): ASoC: SOF: mediatek: revise mt8195 clock sequence ASoC: SOF: mediatek: Add shared_size for mediatek common chip information ASoC: SOF: mediatek: mt8195 modify dram type as non-cache ASoC: SOF: mediatek: mt8195 suspend check dsp idle sound/soc/sof/amd/acp-dsp-offset.h | 2 + sound/soc/sof/amd/acp.c | 36 ++++++++++++++++ sound/soc/sof/amd/acp.h | 4 ++ sound/soc/sof/amd/pci-rn.c | 4 ++ sound/soc/sof/amd/renoir.c | 4 ++ sound/soc/sof/mediatek/adsp_helper.h | 1 + sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 7 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 50 +++++++++++++++------- sound/soc/sof/mediatek/mt8195/mt8195.h | 5 +++ 9 files changed, 97 insertions(+), 16 deletions(-) -- 2.34.1

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[PATCH v2] ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix refcount leak in some error paths. Fixes: 0f83f9296d5c ("ASoC: mediatek: Add machine driver for ALC5650 codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- - update fixes tag. v1 Link: https://lore.kernel.org/r/20220603081308.1332-1-linmq006@gmail.com --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d1c94acb4516..e05f2b0231fe 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -315,12 +316,14 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_platform_node: of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: simple-card-utils: Make asoc_simple_clean_reference() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

asoc_simple_clean_reference() returns zero unconditionally. Letting it return void instead makes it easier to see in the caller that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- include/sound/simple_card_utils.h | 2 +- sound/soc/generic/simple-card-utils.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 8faa649f712b..fe2337fde1f4 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -173,7 +173,7 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link_component *platfo void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus, int is_single_links); -int asoc_simple_clean_reference(struct snd_soc_card *card); +void asoc_simple_clean_reference(struct snd_soc_card *card); void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index da0c27828ce6..a542dec92ad5 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -609,7 +609,7 @@ void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus, } EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_cpu); -int asoc_simple_clean_reference(struct snd_soc_card *card) +void asoc_simple_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; struct snd_soc_dai_link_component *cpu; @@ -622,7 +622,6 @@ int asoc_simple_clean_reference(struct snd_soc_card *card) for_each_link_codecs(dai_link, j, codec) of_node_put(codec->of_node); } - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_clean_reference); @@ -878,7 +877,9 @@ int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_clean_reference(card); + asoc_simple_clean_reference(card); + + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_remove); base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH] ASoC: Intel: broadwell: Make broadwell_disable_jack() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

broadwell_disable_jack() returns zero unconditionally. Letting it return void instead makes it easier to see in the callers that there is no error to handle. This is a preparation for making platform remove callbacks return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/intel/boards/broadwell.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index d37c74fd1a3c..280e84bbe185 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -227,7 +227,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { }, }; -static int broadwell_disable_jack(struct snd_soc_card *card) +static void broadwell_disable_jack(struct snd_soc_card *card) { struct snd_soc_component *component; @@ -239,13 +239,13 @@ static int broadwell_disable_jack(struct snd_soc_card *card) break; } } - - return 0; } static int broadwell_suspend(struct snd_soc_card *card) { - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static int broadwell_resume(struct snd_soc_card *card){ @@ -315,7 +315,9 @@ static int broadwell_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return broadwell_disable_jack(card); + broadwell_disable_jack(card); + + return 0; } static struct platform_driver broadwell_audio = { base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

3 2

[GIT PULL] ASoC fixes for v5.19-rc1
by Mark Brown 07 Jun '22

07 Jun '22

The following changes since commit ab0925ab728163b854ea17da09e299254a86076e: ASoC: da7219: Small fixes for jack detection and removal (2022-05-31 10:16:59 +0200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc1 for you to fetch changes up to 2abdf9f80019e8244d3806ed0e1c9f725e50b452: ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() (2022-06-06 12:31:28 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more fixes for v5.19 which came in during the second half of the merge window, again nothing that's really remarkable outside of the individual drivers. ---------------------------------------------------------------- Adam Ford (1): ASoC: wm8962: Fix suspend while playing music Amadeusz Sławiński (1): ASoC: SOF: Fix potential NULL pointer dereference Charles Keepax (6): ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control Dan Carpenter (2): ASoC: SOF: ipc-msg-injector: Propagate write errors correctly ASoC: SOF: ipc-msg-injector: Fix reversed if statement Hui Wang (1): ASoC: nau8822: Add operation for internal PLL off and on Mark Brown (3): ASoC: Fixup Cirrus SX control usage ASoC: es8328: Fix event generation for deemphasis control ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() Shengjiu Wang (1): ASoC: fsl_sai: Add support for i.MX8MN Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback xliu (1): ASoC: Intel: cirrus-common: fix incorrect channel mapping sound/soc/codecs/cs35l36.c | 3 ++- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 8 +++--- sound/soc/codecs/cs42l56.c | 4 +-- sound/soc/codecs/cs53l30.c | 16 ++++++------ sound/soc/codecs/es8328.c | 5 +++- sound/soc/codecs/nau8822.c | 4 +++ sound/soc/codecs/nau8822.h | 3 +++ sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/fsl/fsl_sai.c | 1 + sound/soc/intel/boards/sof_cirrus_common.c | 40 ++++++++++++++++++++++++++--- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 28 +++++++++++--------- 15 files changed, 86 insertions(+), 35 deletions(-)

2 1

[PATCH] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/G2UL SoC
by Biju Das 07 Jun '22

07 Jun '22

Document RZ/G2U2L SSI bindings. RZ/G2UL SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. While at it add a '.' at the end of dmas description for the first cell. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 7e8d252f7bca..0d9840375132 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -13,6 +13,7 @@ properties: compatible: items: - enum: + - renesas,r9a07g043-ssi # RZ/G2UL - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi @@ -50,7 +51,7 @@ properties: minItems: 1 maxItems: 2 description: - The first cell represents a phandle to dmac + The first cell represents a phandle to dmac. The second cell specifies the encoded MID/RID values of the SSI port connected to the DMA client and the slave channel configuration parameters. -- 2.25.1

4 5

[PATCH 0/3] ASoC: Intel: soc-acpi and machine driver updates
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

Small updates to add initial tables for MeteorLake, SoundWire machine driver support for tests without HDMI and RT1019 for consistency on Chromebooks. Bard Liao (1): ASoC: SOF: Intel: add MeteorLake machines Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: allow HDaudio/HDMI disable Yong Zhi (1): ASoC: Intel: boards: rename RTL1019 compatible driver to rt1019p include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/sof_realtek_common.c | 24 +++++------ sound/soc/intel/boards/sof_realtek_common.h | 6 +-- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 24 +++++++---- sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-mtl-match.c | 41 +++++++++++++++++++ 8 files changed, 75 insertions(+), 26 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-mtl-match.c -- 2.34.1

2 4

[PATCH v5 00/20] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 07 Jun '22

07 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (20): ASoC: mediatek: mt6366: support for mt6366 codec dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add misc driver and register definitions ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/codecs/mt6358.c | 1 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 651 ++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 244 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3009 +++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + .../soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 873 +++++ .../soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++ .../soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1286 +++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 423 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 ++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 698 ++++ .../mediatek/mt8186/mt8186-interconnection.h | 69 + .../soc/mediatek/mt8186/mt8186-misc-control.c | 252 ++ .../mediatek/mt8186/mt8186-mt6366-common.c | 59 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1003 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 979 ++++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 ++++++++++++++++ 32 files changed, 14932 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h -- 2.18.0

3 24

[PATCH 0/2] Switch to use internal PLL for iMCLK
by Hui Wang 07 Jun '22

07 Jun '22

Hi David.Lin, Taking your advice and try to enable internal PLL to get a more accurate sample rate. And I also changed the fsl-asoc-card.c to support the nau8822 codec, now the sound quality is pretty good on my imx6sx EVB. Please help take a look at these 2 patches on codec driver. Thanks. Hui Wang (2): ASoC: nau8822: Add operation for internal PLL off and on ASoC: nau8822: Disable internal PLL if freq_out is zero sound/soc/codecs/nau8822.c | 11 +++++++++++ sound/soc/codecs/nau8822.h | 3 +++ 2 files changed, 14 insertions(+) -- 2.25.1

3 14

[PATCH v5 0/2]Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ v4 --> v5: Resolved: . Used separate firmware files. . Deleted useless initialization variables v3 --> v4: Resolved: . Assigning direct address without (void *) typecast. . Used snd_soc_card_jack_new instead of snd_soc_card_jack_new_pins as per suggestion. v2 --> v3: Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 233 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 282 insertions(+), 16 deletions(-) -- 2.25.1

3 4

[PATCH v3 0/6] OPE support on Tegra210 and later
by Sameer Pujar 07 Jun '22

07 Jun '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v2 -> v3: --------- * Drop "Device Tree Bindings" string from bindind doc titles for OPE, PEQ and MBDRC. v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

3 10

[PATCH 0/4] ASoC: Drop some i2c noop remove callbacks
by Uwe Kleine-König 07 Jun '22

07 Jun '22

From: Uwe Kleine-König <uwe(a)kleine-koenig.org> Hello, I intended to send this after -rc1 was cut, but found a few spare minutes to prepare this series. All four patches were sent already before based on v5.18, but there were some conflicting changes added in the merge window. This series contains the four patches on top of current linus/master and so bases on a tree including the conflicting changes. Expecting no more sound changes in this merge window, this should apply cleanly on top of -rc1. Best regards Uwe Uwe Kleine-König (4): ASoC: ak4642: Drop no-op remove function ASoC: da7219: Drop no-op remove function ASoC: lm49453: Drop no-op remove function ASoC: da732x: Drop no-op remove function sound/soc/codecs/ak4613.c | 6 ------ sound/soc/codecs/da7219.c | 6 ------ sound/soc/codecs/da732x.c | 6 ------ sound/soc/codecs/lm49453.c | 6 ------ 4 files changed, 24 deletions(-) base-commit: 50fd82b3a9a9335df5d50c7ddcb81c81d358c4fc -- 2.36.1

2 5

[PATCH] ASoC: ak4642: Drop no-op remove function
by Uwe Kleine-König 07 Jun '22

07 Jun '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/ak4613.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 034195c83bd7..20a85a81b2ad 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -684,18 +684,12 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, &ak4613_dai, 1); } -static int ak4613_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static struct i2c_driver ak4613_i2c_driver = { .driver = { .name = "ak4613-codec", .of_match_table = ak4613_of_match, }, .probe = ak4613_i2c_probe, - .remove = ak4613_i2c_remove, .id_table = ak4613_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH v4 0/2] Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ v3 --> v4: Resolved: . Assigning direct address without (void *) typecast. . Used snd_soc_card_jack_new instead of snd_soc_card_jack_new_pins as per suggestion. v2 --> v3: Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 235 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 283 insertions(+), 17 deletions(-) -- 2.25.1

4 7

[PATCH v4 00/17] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (16): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 7 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 853 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 37 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 302 +++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 27 +- sound/soc/codecs/cs35l41-lib.c | 82 ++- sound/soc/codecs/cs35l41.c | 71 +-- 14 files changed, 1362 insertions(+), 74 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

6 34

[PATCH 00/16] ASoC: trivial changes for cppcheck warnings
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

This can probably wait for the next merge window, I found a number of cppcheck warnings that I didn't see in my last checks. The irony is that the only really important issue found by cppcheck was on one of my previous DPCM changes (submitted separately as a fix). Pierre-Louis Bossart (16): ASoC: cs35l45: typo in argument definition ASoC: cs42l42: remove redundant test ASoC: wcd-mbhc-v2: remove useless initialization ASoC: wcd9335: remove redundant tests ASoC: Intel: atom: sst: remove useless initialization ASoC: Intel: atom: sst_ipc: remove redundant test ASoC: Intel: atom: sst_ipc: remove useless initializations ASoC: Intel: atom: controls: remove useless initializations ASoC: Intel: boards: reset acpi_chan_package ASoC: Intel: sof_pcm512x: remove unnecessary init ASoC: mediatek: mt8195: simplify error handling ASoC: qcom: q6dsp: q6adm: remove useless initializations ASoC: qcom: q6dsp: remove spurious space ASoC: rockchip: simplify error handling ASoC: samsung: snow: simplify error handling ASoC: meson: remove useless initialization sound/soc/codecs/cs35l45.h | 2 +- sound/soc/codecs/cs42l42.c | 3 +-- sound/soc/codecs/wcd-mbhc-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 4 ++-- sound/soc/intel/atom/sst-atom-controls.c | 4 ++-- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_ipc.c | 8 ++++---- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 8 ++++---- sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/qcom/qdsp6/q6adm.c | 6 +++--- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 ++-- sound/soc/samsung/snow.c | 2 +- 16 files changed, 27 insertions(+), 28 deletions(-) -- 2.30.2

2 17

[PATCH v3 0/2] Add Machine driver support for nau8825, max98560 and rt5682s, rt1019
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Resolved compilation error: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c: In function ‘acp_card_nau8825_init’: /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:479:15: error: too many arguments to function ‘snd_soc_card_jack_new’ 479 | ret = snd_soc_card_jack_new(card, "Headset Jack", | ^~~~~~~~~~~~~~~~~~~~~ In file included from /build/stage/linux/include/sound/soc.h:1362, from /build/stage/linux/sound/soc/amd/acp/acp-mach-common.c:20: /build/stage/linux/include/sound/soc-card.h:18:5: note: declared here 18 | int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 235 ++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 283 insertions(+), 17 deletions(-) -- 2.25.1

5 7

[PATCH v2 0/2] ASOC: amd: acp: Add Machine driver changes for
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 276 insertions(+), 17 deletions(-) -- 2.25.1

2 4

[PATCH v3 00/17] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (16): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 7 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 815 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 37 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 302 ++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 27 +- sound/soc/codecs/cs35l41-lib.c | 82 +++- sound/soc/codecs/cs35l41.c | 71 +-- 14 files changed, 1324 insertions(+), 74 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

3 23

[PATCH v2 00/26] *ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

4 34

[PATCH 00/26] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ALSA: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.32.0

5 37

[PATCH] ASoC: es8328: Fix event generation for deemphasis control
by Mark Brown 07 Jun '22

07 Jun '22

Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.30.2

1 1

[PATCH] ASoC: wm8962: Fix suspend while playing music
by Adam Ford 07 Jun '22

07 Jun '22

If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34cd5a2a997c..5cca89364280 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3868,6 +3868,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.25.1

3 2

[PATCH] ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/ab8500-codec.c | 2 -- sound/soc/codecs/ab8500-codec.h | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index aefafb0b7b97..cbd4a92cb06c 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -12,8 +12,6 @@ * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #include <linux/kernel.h> diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 0ac87d0446c2..2a6f6409f1f8 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -11,8 +11,6 @@ * Mikko J. Lehto <mikko.lehto(a)symbio.com>, * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #ifndef AB8500_CODEC_REGISTERS_H -- 2.34.1

2 1

[PATCH] ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/ux500/mop500.c | 2 -- sound/soc/ux500/mop500_ab8500.c | 2 -- sound/soc/ux500/mop500_ab8500.h | 2 -- sound/soc/ux500/ux500_msp_dai.c | 2 -- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 2 -- sound/soc/ux500/ux500_msp_i2s.h | 2 -- sound/soc/ux500/ux500_pcm.c | 2 -- sound/soc/ux500/ux500_pcm.h | 2 -- 9 files changed, 18 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4f41bb0ab2b0..fdd55d772b8e 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -4,8 +4,6 @@ * * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) * for ST-Ericsson. - * - * License terms: */ #include <asm/mach-types.h> diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 1ea1729984a9..e5e73a2bd9fe 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h index 087ef246d87d..98de80a9cc4f 100644 --- a/sound/soc/ux500/mop500_ab8500.h +++ b/sound/soc/ux500/mop500_ab8500.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef MOP500_AB8500_H diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 21052378a32e..56532b62faf3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index fcd4b26f5d2d..30bf70838196 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_msp_dai_H diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index fd0b88bb7921..d113411a19f8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -6,8 +6,6 @@ * Roger Nilsson <roger.xr.nilsson(a)stericsson.com>, * Sandeep Kaushik <sandeep.kaushik(a)st.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 756b3973af9a..d45b5e2831cc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * for ST-Ericsson. - * - * License terms: */ diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 18191084b8b8..d3802e5ef196 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <asm/page.h> diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index ff3ef7223db6..bd4348ebf9a1 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_PCM_H #define UX500_PCM_H -- 2.34.1

2 1

[PATCH] ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec()
by Kuninori Morimoto 07 Jun '22

07 Jun '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 95cfc0a0aaf5752071 ("ASoC: simple-card: Add support for codec2codec DAI links") added the function asoc_simple_init_dai_link_params() to initialize dai_link "params". It is very straight naming, but difficult to noticed that it is for Codec2Codec support. Handling Codec2Codec is one of very tricky part on ALSA SoC, thus it is very important to clarify it. This patch renames the function name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 539d7f081bd7..fa080f166345 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -513,7 +513,7 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } -static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, +static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct simple_dai_props *dai_props) { struct snd_soc_dai_link *dai_link = rtd->dai_link; @@ -575,7 +575,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = asoc_simple_init_dai_link_params(rtd, props); + ret = asoc_simple_init_for_codec2codec(rtd, props); if (ret < 0) return ret; -- 2.25.1

2 1

[PATCH] ASoC: fsl_sai: Add support for i.MX8MN
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SAI module on i.MX8MN is almost same as i.MX8MP, So reuse same soc data as i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index dae16a14f177..59a91a6d91a9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1299,6 +1299,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx8mm-sai", .data = &fsl_sai_imx8mm_data }, { .compatible = "fsl,imx8mp-sai", .data = &fsl_sai_imx8mp_data }, { .compatible = "fsl,imx8ulp-sai", .data = &fsl_sai_imx8ulp_data }, + { .compatible = "fsl,imx8mn-sai", .data = &fsl_sai_imx8mp_data }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 2.17.1

2 1

[PATCH] ASoC: codecs: da7210: add check for i2c_add_driver
by Jiasheng Jiang 07 Jun '22

07 Jun '22

As i2c_add_driver could return error if fails, it should be better to check the return value. However, if the CONFIG_I2C and CONFIG_SPI_MASTER are both true, the return value of i2c_add_driver will be covered by spi_register_driver. Therefore, it is necessary to add check and return error if fails. Fixes: aa0e25caafb7 ("ASoC: da7210: Add support for spi regmap") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/da7210.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3fa3042e4424..76a21976ccdd 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1335,6 +1335,8 @@ static int __init da7210_modinit(void) int ret = 0; #if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&da7210_i2c_driver); + if (ret) + return ret; #endif #if defined(CONFIG_SPI_MASTER) ret = spi_register_driver(&da7210_spi_driver); -- 2.25.1

2 1

[PATCH 1/3] ASoC: fsl_sai: use local device pointer
by Marco Felsch 07 Jun '22

07 Jun '22

Use a local variable to dereference the device pointer once and use the local variable in further calls. No functional changes. Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 53 +++++++++++++++++++++-------------------- 1 file changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..a7637d602f3c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1004,6 +1004,7 @@ static int fsl_sai_runtime_resume(struct device *dev); static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; struct fsl_sai *sai; struct regmap *gpr; struct resource *res; @@ -1012,12 +1013,12 @@ static int fsl_sai_probe(struct platform_device *pdev) int irq, ret, i; int index; - sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); + sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; sai->pdev = pdev; - sai->soc_data = of_device_get_match_data(&pdev->dev); + sai->soc_data = of_device_get_match_data(dev); sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -1032,18 +1033,18 @@ static int fsl_sai_probe(struct platform_device *pdev) ARRAY_SIZE(fsl_sai_reg_defaults_ofs8); } - sai->regmap = devm_regmap_init_mmio(&pdev->dev, base, &fsl_sai_regmap_config); + sai->regmap = devm_regmap_init_mmio(dev, base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { - dev_err(&pdev->dev, "regmap init failed\n"); + dev_err(dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } - sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + sai->bus_clk = devm_clk_get(dev, "bus"); /* Compatible with old DTB cases */ if (IS_ERR(sai->bus_clk) && PTR_ERR(sai->bus_clk) != -EPROBE_DEFER) - sai->bus_clk = devm_clk_get(&pdev->dev, "sai"); + sai->bus_clk = devm_clk_get(dev, "sai"); if (IS_ERR(sai->bus_clk)) { - dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + dev_err(dev, "failed to get bus clock: %ld\n", PTR_ERR(sai->bus_clk)); /* -EPROBE_DEFER */ return PTR_ERR(sai->bus_clk); @@ -1051,9 +1052,9 @@ static int fsl_sai_probe(struct platform_device *pdev) for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); - sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { - dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + dev_err(dev, "failed to get mclk%d clock: %ld\n", i + 1, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } @@ -1068,10 +1069,10 @@ static int fsl_sai_probe(struct platform_device *pdev) if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, fsl_sai_isr, IRQF_SHARED, + ret = devm_request_irq(dev, irq, fsl_sai_isr, IRQF_SHARED, np->name, sai); if (ret) { - dev_err(&pdev->dev, "failed to claim irq %u\n", irq); + dev_err(dev, "failed to claim irq %u\n", irq); return ret; } @@ -1088,7 +1089,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && of_find_property(np, "fsl,sai-asynchronous", NULL)) { /* error out if both synchronous and asynchronous are present */ - dev_err(&pdev->dev, "invalid binding for synchronous mode\n"); + dev_err(dev, "invalid binding for synchronous mode\n"); return -EINVAL; } @@ -1109,7 +1110,7 @@ static int fsl_sai_probe(struct platform_device *pdev) of_device_is_compatible(np, "fsl,imx6ul-sai")) { gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); if (IS_ERR(gpr)) { - dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + dev_err(dev, "cannot find iomuxc registers\n"); return PTR_ERR(gpr); } @@ -1127,23 +1128,23 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; platform_set_drvdata(pdev, sai); - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) { - ret = fsl_sai_runtime_resume(&pdev->dev); + pm_runtime_enable(dev); + if (!pm_runtime_enabled(dev)) { + ret = fsl_sai_runtime_resume(dev); if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_get_sync(dev); if (ret < 0) { - pm_runtime_put_noidle(&pdev->dev); + pm_runtime_put_noidle(dev); goto err_pm_get_sync; } /* Get sai version */ - ret = fsl_sai_check_version(&pdev->dev); + ret = fsl_sai_check_version(dev); if (ret < 0) - dev_warn(&pdev->dev, "Error reading SAI version: %d\n", ret); + dev_warn(dev, "Error reading SAI version: %d\n", ret); /* Select MCLK direction */ if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && @@ -1152,7 +1153,7 @@ static int fsl_sai_probe(struct platform_device *pdev) FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); } - ret = pm_runtime_put_sync(&pdev->dev); + ret = pm_runtime_put_sync(dev); if (ret < 0) goto err_pm_get_sync; @@ -1165,12 +1166,12 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) goto err_pm_get_sync; } else { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) goto err_pm_get_sync; } - ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, + ret = devm_snd_soc_register_component(dev, &fsl_component, &sai->cpu_dai_drv, 1); if (ret) goto err_pm_get_sync; @@ -1178,10 +1179,10 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; err_pm_get_sync: - if (!pm_runtime_status_suspended(&pdev->dev)) - fsl_sai_runtime_suspend(&pdev->dev); + if (!pm_runtime_status_suspended(dev)) + fsl_sai_runtime_suspend(dev); err_pm_disable: - pm_runtime_disable(&pdev->dev); + pm_runtime_disable(dev); return ret; } -- 2.30.2

3 6

[PATCH] ASoC: mediatek: mt8173: Fix refcount leak in mt8173_rt5650_rt5676_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix missing of_node_put() in error paths. Fixes: 94319ba10eca ("ASoC: mediatek: Use platform_of_node for machine drivers") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 70bf312e855f..8794720cea3a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -256,14 +256,16 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); if (!mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } mt8173_rt5650_rt5676_codec_conf[0].dlc.of_node = mt8173_rt5650_rt5676_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node; @@ -276,13 +278,15 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_node: of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: mediatek: remove unnecessary check of clk_disable_unprepare
by cgel.zte＠gmail.com 07 Jun '22

07 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 31494930433f..dcaeeeb8aac7 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -286,10 +286,8 @@ static int mt8173_afe_dais_set_clks(struct mtk_base_afe *afe, static void mt8173_afe_dais_disable_clks(struct mtk_base_afe *afe, struct clk *m_ck, struct clk *b_ck) { - if (m_ck) - clk_disable_unprepare(m_ck); - if (b_ck) - clk_disable_unprepare(b_ck); + clk_disable_unprepare(m_ck); + clk_disable_unprepare(b_ck); } static int mt8173_afe_i2s_startup(struct snd_pcm_substream *substream, -- 2.25.1

2 1

[PATCH V4] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 07 Jun '22

07 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- v2: - Add remove to put PM usage counter. v3: - Modify the commit message. --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

3 2

[PATCH 1/2] ASoC: SOF: ipc-msg-injector: Propagate write errors correctly
by Dan Carpenter 07 Jun '22

07 Jun '22

This code is supposed to propagate errors from simple_write_to_buffer() or return -EFAULT if "size != count". However "size" needs to be signed for the code to work correctly and the case where "size == 0" is not handled correctly. Fixes: 066c67624d8c ("ASoC: SOF: ipc-msg-injector: Add support for IPC4 messages") Fixes: 2f0b1b013bbc ("ASoC: SOF: debug: Add support for IPC message injection") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/sof-client-ipc-msg-injector.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index 03490a4d4ae7..030cb97d7713 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -150,7 +150,7 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu { struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -158,8 +158,10 @@ static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *bu size = simple_write_to_buffer(priv->tx_buffer, priv->max_msg_size, ppos, buffer, count); + if (size < 0) + return size; if (size != count) - return size > 0 ? -EFAULT : size; + return -EFAULT; memset(priv->rx_buffer, 0, priv->max_msg_size); @@ -179,7 +181,7 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, struct sof_client_dev *cdev = file->private_data; struct sof_msg_inject_priv *priv = cdev->data; struct sof_ipc4_msg *ipc4_msg = priv->tx_buffer; - size_t size; + ssize_t size; int ret; if (*ppos) @@ -192,8 +194,10 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, size = simple_write_to_buffer(&ipc4_msg->header_u64, sizeof(ipc4_msg->header_u64), ppos, buffer, count); + if (size < 0) + return size; if (size != sizeof(ipc4_msg->header_u64)) - return size > 0 ? -EFAULT : size; + return -EFAULT; count -= size; if (!count) { @@ -201,8 +205,10 @@ static ssize_t sof_msg_inject_ipc4_dfs_write(struct file *file, size = simple_write_to_buffer(ipc4_msg->data_ptr, priv->max_msg_size, ppos, buffer, count); + if (size < 0) + return size; if (size != count) - return size > 0 ? -EFAULT : size; + return -EFAULT; } ipc4_msg->data_size = count; -- 2.35.1

3 4

[PATCH] ASoC: ops: Remove unneeded delay.h inclusion
by Mark Brown 07 Jun '22

07 Jun '22

The ops code does not do any sleeps or delays so does not need delay.h. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 0d373435ceb8..2d39370ddeca 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -14,7 +14,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> -#include <linux/delay.h> #include <linux/pm.h> #include <linux/bitops.h> #include <linux/ctype.h> -- 2.30.2

1 1

[PATCH] ASoC: ops: Clarify snd_soc_info_volsw_sx()
by Mark Brown 07 Jun '22

07 Jun '22

Currently snd_soc_info_volsw_sx() is implemented indirectly, wrapping snd_soc_info_volsw() and modifying the values it sets up rather than directly setting up the values reported to userspace. This makes it much harder to follow what the intended behaviour of these controls is. Let's rewrite the function to be self contained with a clarifying comment at the top in an effort to help maintainability. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Build tested only. sound/soc/soc-ops.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a0ca58ba1627..03010577046c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -203,7 +203,8 @@ EXPORT_SYMBOL_GPL(snd_soc_info_volsw); * Callback to provide information about a single mixer control, or a double * mixer control that spans 2 registers of the SX TLV type. SX TLV controls * have a range that represents both positive and negative values either side - * of zero but without a sign bit. + * of zero but without a sign bit. min is the minimum register value, max is + * the number of steps. * * Returns 0 for success. */ @@ -212,12 +213,21 @@ int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int max; - snd_soc_info_volsw(kcontrol, uinfo); - /* Max represents the number of levels in an SX control not the - * maximum value, so add the minimum value back on - */ - uinfo->value.integer.max += mc->min; + if (mc->platform_max) + max = mc->platform_max; + else + max = mc->max; + + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max; return 0; } -- 2.30.2

2 3

[PATCH] ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: format:b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/cros_ec_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8b0a9c788a26..11e7b3f6d410 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -995,6 +995,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n", priv->ap_shm_phys_addr, priv->ap_shm_len); } + of_node_put(node); } #endif -- 2.25.1

2 1

[PATCH] ASoC: max9860: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

As part of moving to remove the old style defines for the bus clocks update the max9860 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9860.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 82f20a8e27ad..a1d0179e12c7 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -448,9 +448,9 @@ static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; struct max9860_priv *max9860 = snd_soc_component_get_drvdata(component); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + case SND_SOC_DAIFMT_CBC_CFC: max9860->fmt = fmt; return 0; -- 2.30.2

2 2

[PATCH] ASoC: tfa9879: Use modern ASoC DAI format terminology
by Mark Brown 07 Jun '22

07 Jun '22

As part of moving to remove the old style defines for the bus clocks update the tfa9879 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tfa9879.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 3d8e8c2276f0..41a9b1b76e62 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -111,8 +111,8 @@ static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) int i2s_set; int sck_pol; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.30.2

2 2

[PATCH] ASoC: SOF: Fix potential NULL pointer dereference
by Amadeusz Sławiński 07 Jun '22

07 Jun '22

Cleanup path for sof_prepare_widgets_in_path() should check if unprepare callback exists before calling it, instead it checks if it does not exist. Fix the check. Fixes: 370014c8197a ("ASoC: SOF: Add a prepare op to IPC topology widget ops") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index cd41d0599cef..789d0a1f1a77 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -323,7 +323,7 @@ sof_prepare_widgets_in_path(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget p->walking = false; if (ret < 0) { /* unprepare the source widget */ - if (!widget_ops[widget->id].ipc_unprepare && swidget->prepared) { + if (widget_ops[widget->id].ipc_unprepare && swidget->prepared) { widget_ops[widget->id].ipc_unprepare(swidget); swidget->prepared = false; } -- 2.25.1

4 4

[PATCH 1/2] ASoC: codecs: tfa989x: Add support for tfa9890
by Alexander Martinz 07 Jun '22

07 Jun '22

The initialization sequence is taken from the version provided by the supplier [1]. This allows speakers using the TFA9890 amplifier to work, which are used by various mobile phones such as the SHIFT6mq. [1]: https://source.codeaurora.org/external/mas/tfa98xx/tree/src/tfa_init.c?id=d… Signed-off-by: Alexander Martinz <amartinz(a)shiftphones.com> --- sound/soc/codecs/tfa989x.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/tfa989x.c b/sound/soc/codecs/tfa989x.c index dc86852752c5..8ab2656de750 100644 --- a/sound/soc/codecs/tfa989x.c +++ b/sound/soc/codecs/tfa989x.c @@ -40,12 +40,14 @@ #define TFA989X_I2S_SEL_REG 0x0a #define TFA989X_I2S_SEL_REG_SPKR_MSK GENMASK(10, 9) /* speaker impedance */ #define TFA989X_I2S_SEL_REG_DCFG_MSK GENMASK(14, 11) /* DCDC compensation */ +#define TFA989X_HIDE_UNHIDE_KEY 0x40 #define TFA989X_PWM_CONTROL 0x41 #define TFA989X_CURRENTSENSE1 0x46 #define TFA989X_CURRENTSENSE2 0x47 #define TFA989X_CURRENTSENSE3 0x48 #define TFA989X_CURRENTSENSE4 0x49 +#define TFA9890_REVISION 0x80 #define TFA9895_REVISION 0x12 #define TFA9897_REVISION 0x97 @@ -188,6 +190,33 @@ static struct snd_soc_dai_driver tfa989x_dai = { .ops = &tfa989x_dai_ops, }; +static int tfa9890_init(struct regmap *regmap) +{ + int ret; + + /* unhide keys to allow updating them */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x5a6b); + if (ret) + return ret; + + /* update PLL registers */ + ret = regmap_set_bits(regmap, 0x59, 0x3); + if (ret) + return ret; + + /* hide keys again */ + ret = regmap_write(regmap, TFA989X_HIDE_UNHIDE_KEY, 0x0000); + if (ret) + return ret; + + return regmap_write(regmap, TFA989X_CURRENTSENSE2, 0x7BE1); +} + +static const struct tfa989x_rev tfa9890_rev = { + .rev = TFA9890_REVISION, + .init = tfa9890_init, +}; + static const struct reg_sequence tfa9895_reg_init[] = { /* some other registers must be set for optimal amplifier behaviour */ { TFA989X_BAT_PROT, 0x13ab }, @@ -376,6 +405,7 @@ static int tfa989x_i2c_probe(struct i2c_client *i2c) } static const struct of_device_id tfa989x_of_match[] = { + { .compatible = "nxp,tfa9890", .data = &tfa9890_rev }, { .compatible = "nxp,tfa9895", .data = &tfa9895_rev }, { .compatible = "nxp,tfa9897", .data = &tfa9897_rev }, { } -- 2.36.1

4 5

[PATCH] ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: format:f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.25.1

2 1

[PATCH] ASoC: nau8822: Don't reconfigure PLL to the same values
by Mark Brown 07 Jun '22

07 Jun '22

When we configure the PLL record the input and output frequency, then if we get asked to configure the same values again just skip reprogramming the hardware. This makes things a bit easier to use for machine drivers since it means they don't need to keep track of if they've programmed the PLL so much. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8822.c | 7 +++++++ sound/soc/codecs/nau8822.h | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 08f6c56dc387..f4f68b549e1a 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -726,6 +726,10 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, struct nau8822_pll *pll_param = &nau8822->pll; int ret, fs; + if (freq_in == pll_param->freq_in && + freq_out == pll_param->freq_out) + return 0; + fs = freq_out / 256; ret = nau8822_calc_pll(freq_in, fs, pll_param); @@ -762,6 +766,9 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, snd_soc_component_update_bits(component, NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); + pll_param->freq_in = freq_in; + pll_param->freq_out = freq_out; + return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index b45d42c15de6..547ec057f853 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -198,6 +198,8 @@ struct nau8822_pll { int mclk_scaler; int pll_frac; int pll_int; + int freq_in; + int freq_out; }; /* Codec Private Data */ -- 2.30.2

1 1

[PATCH v2] ASoC: samsung: Fix error handling in aries_audio_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_get_child_by_name() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. This function is missing of_node_put(cpu) in the error path. Fix this by goto out label. of_node_put() will check NULL pointer. Fixes: 7a3a7671fa6c ("ASoC: samsung: Add driver for Aries boards") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag. v1 Link: https://lore.kernel.org/r/20220602130218.14901-1-linmq006@gmail.com --- sound/soc/samsung/aries_wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index bb0cf4244e00..edee02d7f100 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -628,8 +628,10 @@ static int aries_audio_probe(struct platform_device *pdev) return -EINVAL; codec = of_get_child_by_name(dev->of_node, "codec"); - if (!codec) - return -EINVAL; + if (!codec) { + ret = -EINVAL; + goto out; + } for_each_card_prelinks(card, i, dai_link) { dai_link->codecs->of_node = of_parse_phandle(codec, -- 2.25.1

3 2

[PATCH v2] ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe
by Miaoqian Lin 07 Jun '22

07 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag v1 Link: https://lore.kernel.org/r/20220602114529.6303-1-linmq006@gmail.com --- sound/soc/codecs/cros_ec_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8b0a9c788a26..11e7b3f6d410 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -995,6 +995,7 @@ static int cros_ec_codec_platform_probe(struct platform_device *pdev) dev_dbg(dev, "ap_shm_phys_addr=%#llx len=%#x\n", priv->ap_shm_phys_addr, priv->ap_shm_len); } + of_node_put(node); } #endif -- 2.25.1

4 3

[PATCH] ASoC: max98390: use linux/gpio/consumer.h to fix build
by Randy Dunlap 07 Jun '22

07 Jun '22

Change the header file to fix build errors in max98390.c: ../sound/soc/codecs/max98390.c: In function 'max98390_i2c_probe': ../sound/soc/codecs/max98390.c:1076:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 1076 | reset_gpio = devm_gpiod_get_optional(&i2c->dev, ../sound/soc/codecs/max98390.c:1077:55: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 1077 | "reset", GPIOD_OUT_HIGH); ../sound/soc/codecs/max98390.c:1077:55: note: each undeclared identifier is reported only once for each function it appears in ../sound/soc/codecs/max98390.c:1083:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1083 | gpiod_set_value_cansleep(reset_gpio, 0); Fixes: 397ff0249606 ("ASoC: max98390: Add reset gpio control") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Steve Lee <steve.lee.analog(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -10,7 +10,7 @@ #include <linux/cdev.h> #include <linux/dmi.h> #include <linux/firmware.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_gpio.h>

2 1

[PATCH] ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put()
by Mark Brown 07 Jun '22

07 Jun '22

Currently wm_adsp_fw_put() returns 0 rather than 1 when updating the value of the control, meaning that no event is generated to userspace. Fix this by setting the default return value to 1, the code already exits early with a return value of 0 if the value is unchanged. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..6d7fd88243aa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -333,7 +333,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; -- 2.30.2

2 2

[PATCH] ASoC: imx-audmux: Silence a clang warning
by Fabio Estevam 07 Jun '22

07 Jun '22

Change the of_device_get_match_data() cast to (uintptr_t) to silence the following clang warning: sound/soc/fsl/imx-audmux.c:301:16: warning: cast to smaller integer type 'enum imx_audmux_type' from 'const void *' [-Wvoid-pointer-to-enum-cast] Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 6a8b8b582db1 ("ASoC: imx-audmux: Remove unused .id_table") Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..a8e5e0f57faf 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -298,7 +298,7 @@ static int imx_audmux_probe(struct platform_device *pdev) audmux_clk = NULL; } - audmux_type = (enum imx_audmux_type)of_device_get_match_data(&pdev->dev); + audmux_type = (uintptr_t)of_device_get_match_data(&pdev->dev); switch (audmux_type) { case IMX31_AUDMUX: -- 2.25.1

3 2

[PATCH] ASoC: fsl_mqs: simplify the code with adding fsl_mqs_soc_data
by Shengjiu Wang 07 Jun '22

07 Jun '22

Add soc specific data struct fsl_mqs_soc_data, move the definition of control register, each function bits to it, then the code can be simplified. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_mqs.c | 119 ++++++++++++++++++++++++---------------- 1 file changed, 71 insertions(+), 48 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index ceaecbe3a25e..8a8d727319d6 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -10,6 +10,7 @@ #include <linux/moduleparam.h> #include <linux/mfd/syscon.h> #include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> +#include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/pm.h> @@ -29,15 +30,41 @@ #define MQS_CLK_DIV_MASK (0xFF << 0) #define MQS_CLK_DIV_SHIFT (0) +/** + * struct fsl_mqs_soc_data - soc specific data + * + * @use_gpr: control register is in General Purpose Register group + * @ctrl_off: control register offset + * @en_mask: enable bit mask + * @en_shift: enable bit shift + * @rst_mask: reset bit mask + * @rst_shift: reset bit shift + * @osr_mask: oversample bit mask + * @osr_shift: oversample bit shift + * @div_mask: clock divider mask + * @div_shift: clock divider bit shift + */ +struct fsl_mqs_soc_data { + bool use_gpr; + int ctrl_off; + int en_mask; + int en_shift; + int rst_mask; + int rst_shift; + int osr_mask; + int osr_shift; + int div_mask; + int div_shift; +}; + /* codec private data */ struct fsl_mqs { struct regmap *regmap; struct clk *mclk; struct clk *ipg; + const struct fsl_mqs_soc_data *soc; - unsigned int reg_iomuxc_gpr2; unsigned int reg_mqs_ctrl; - bool use_gpr; }; #define FSL_MQS_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) @@ -65,19 +92,11 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, res = mclk_rate % (32 * lrclk * 2 * 8); if (res == 0 && div > 0 && div <= 256) { - if (mqs_priv->use_gpr) { - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_CLK_DIV_MASK, - (div - 1) << IMX6SX_GPR2_MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, 0); - } else { - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_CLK_DIV_MASK, - (div - 1) << MQS_CLK_DIV_SHIFT); - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_OVERSAMPLE_MASK, 0); - } + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->div_mask, + (div - 1) << mqs_priv->soc->div_shift); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->osr_mask, 0); } else { dev_err(component->dev, "can't get proper divider\n"); } @@ -118,14 +137,9 @@ static int fsl_mqs_startup(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, - 1 << IMX6SX_GPR2_MQS_EN_SHIFT); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, - 1 << MQS_EN_SHIFT); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, + 1 << mqs_priv->soc->en_shift); return 0; } @@ -135,12 +149,8 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); - if (mqs_priv->use_gpr) - regmap_update_bits(mqs_priv->regmap, IOMUXC_GPR2, - IMX6SX_GPR2_MQS_EN_MASK, 0); - else - regmap_update_bits(mqs_priv->regmap, REG_MQS_CTRL, - MQS_EN_MASK, 0); + regmap_update_bits(mqs_priv->regmap, mqs_priv->soc->ctrl_off, + mqs_priv->soc->en_mask, 0); } static const struct snd_soc_component_driver soc_codec_fsl_mqs = { @@ -191,12 +201,9 @@ static int fsl_mqs_probe(struct platform_device *pdev) * But in i.MX8QM/i.MX8QXP the control register is moved * to its own domain. */ - if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) - mqs_priv->use_gpr = false; - else - mqs_priv->use_gpr = true; + mqs_priv->soc = of_device_get_match_data(&pdev->dev); - if (mqs_priv->use_gpr) { + if (mqs_priv->soc->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); @@ -280,12 +287,7 @@ static int fsl_mqs_runtime_resume(struct device *dev) return ret; } - if (mqs_priv->use_gpr) - regmap_write(mqs_priv->regmap, IOMUXC_GPR2, - mqs_priv->reg_iomuxc_gpr2); - else - regmap_write(mqs_priv->regmap, REG_MQS_CTRL, - mqs_priv->reg_mqs_ctrl); + regmap_write(mqs_priv->regmap, mqs_priv->soc->ctrl_off, mqs_priv->reg_mqs_ctrl); return 0; } @@ -293,12 +295,7 @@ static int fsl_mqs_runtime_suspend(struct device *dev) { struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); - if (mqs_priv->use_gpr) - regmap_read(mqs_priv->regmap, IOMUXC_GPR2, - &mqs_priv->reg_iomuxc_gpr2); - else - regmap_read(mqs_priv->regmap, REG_MQS_CTRL, - &mqs_priv->reg_mqs_ctrl); + regmap_read(mqs_priv->regmap, mqs_priv->soc->ctrl_off, &mqs_priv->reg_mqs_ctrl); clk_disable_unprepare(mqs_priv->mclk); clk_disable_unprepare(mqs_priv->ipg); @@ -315,9 +312,35 @@ static const struct dev_pm_ops fsl_mqs_pm_ops = { pm_runtime_force_resume) }; +static const struct fsl_mqs_soc_data fsl_mqs_imx8qm_data = { + .use_gpr = false, + .ctrl_off = REG_MQS_CTRL, + .en_mask = MQS_EN_MASK, + .en_shift = MQS_EN_SHIFT, + .rst_mask = MQS_SW_RST_MASK, + .rst_shift = MQS_SW_RST_SHIFT, + .osr_mask = MQS_OVERSAMPLE_MASK, + .osr_shift = MQS_OVERSAMPLE_SHIFT, + .div_mask = MQS_CLK_DIV_MASK, + .div_shift = MQS_CLK_DIV_SHIFT, +}; + +static const struct fsl_mqs_soc_data fsl_mqs_imx6sx_data = { + .use_gpr = true, + .ctrl_off = IOMUXC_GPR2, + .en_mask = IMX6SX_GPR2_MQS_EN_MASK, + .en_shift = IMX6SX_GPR2_MQS_EN_SHIFT, + .rst_mask = IMX6SX_GPR2_MQS_SW_RST_MASK, + .rst_shift = IMX6SX_GPR2_MQS_SW_RST_SHIFT, + .osr_mask = IMX6SX_GPR2_MQS_OVERSAMPLE_MASK, + .osr_shift = IMX6SX_GPR2_MQS_OVERSAMPLE_SHIFT, + .div_mask = IMX6SX_GPR2_MQS_CLK_DIV_MASK, + .div_shift = IMX6SX_GPR2_MQS_CLK_DIV_SHIFT, +}; + static const struct of_device_id fsl_mqs_dt_ids[] = { - { .compatible = "fsl,imx8qm-mqs", }, - { .compatible = "fsl,imx6sx-mqs", }, + { .compatible = "fsl,imx8qm-mqs", .data = &fsl_mqs_imx8qm_data }, + { .compatible = "fsl,imx6sx-mqs", .data = &fsl_mqs_imx6sx_data }, {} }; MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); -- 2.17.1

2 1

[PATCH] ASoC: cs35l45: Make bmc150_magn_remove() return void
by Uwe Kleine-König 07 Jun '22

07 Jun '22

cs35l45_remove() always returns zero. Make it return no value which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c driver remove callbacks is ignored anyway. This prepares making i2c remove callbacks return void, too. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/cs35l45-i2c.c | 4 +++- sound/soc/codecs/cs35l45.c | 4 +--- sound/soc/codecs/cs35l45.h | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 18ebf1e14b91..53a57320d0ab 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -40,7 +40,9 @@ static int cs35l45_i2c_remove(struct i2c_client *client) { struct cs35l45_private *cs35l45 = i2c_get_clientdata(client); - return cs35l45_remove(cs35l45); + cs35l45_remove(cs35l45); + + return 0; } static const struct of_device_id cs35l45_of_match[] = { diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 7822626d3693..64a3abe14010 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -669,7 +669,7 @@ int cs35l45_probe(struct cs35l45_private *cs35l45) } EXPORT_SYMBOL_GPL(cs35l45_probe); -int cs35l45_remove(struct cs35l45_private *cs35l45) +void cs35l45_remove(struct cs35l45_private *cs35l45) { pm_runtime_disable(cs35l45->dev); @@ -677,8 +677,6 @@ int cs35l45_remove(struct cs35l45_private *cs35l45) regulator_disable(cs35l45->vdd_a); /* VDD_BATT must be the last to power-off */ regulator_disable(cs35l45->vdd_batt); - - return 0; } EXPORT_SYMBOL_GPL(cs35l45_remove); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 4e266d19cd1c..680891bcfce9 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -212,6 +212,6 @@ extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs43l45); unsigned int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); -int cs35l45_remove(struct cs35l45_private *cs35l45); +void cs35l45_remove(struct cs35l45_private *cs35l45); #endif /* CS35L45_H */ base-commit: 0d463d016000d68d7e982720b5e4380b2d83409a -- 2.35.1

4 5

[PATCH] ASoC: amd: acp: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/amd/acp/acp-pdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index 424c6e0bb9d6..7a0b26a30051 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -174,7 +174,7 @@ static void acp_dmic_dai_shutdown(struct snd_pcm_substream *substream, struct acp_dev_data *adata = dev_get_drvdata(dev); u32 ext_int_ctrl; - /* Disable DMIC interrrupts */ + /* Disable DMIC interrupts */ ext_int_ctrl = readl(adata->acp_base + ACP_EXTERNAL_INTR_CNTL); ext_int_ctrl |= ~PDM_DMA_INTR_MASK; writel(ext_int_ctrl, adata->acp_base + ACP_EXTERNAL_INTR_CNTL);

2 1

[PATCH] ASoC: stm32: dfsdm: fix typo in comment
by Julia Lawall 07 Jun '22

07 Jun '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/soc/stm/stm32_adfsdm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index 6ee714542b84..122805160e70 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -296,7 +296,7 @@ static int stm32_adfsdm_pcm_new(struct snd_soc_component *component, static int stm32_adfsdm_dummy_cb(const void *data, void *private) { /* - * This dummmy callback is requested by iio_channel_get_all_cb() API, + * This dummy callback is requested by iio_channel_get_all_cb() API, * but the stm32_dfsdm_get_buff_cb() API is used instead, to optimize * DMA transfers. */

2 1

[PATCH v2] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 07 Jun '22

07 Jun '22

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- This patch set depends on: -- Clock driver patches for CGCR reset control support. --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=638002 --- https://patchwork.kernel.org/project/linux-arm-msm/list/?series=637998 Changes since v1: -- Add audio cgcr reset control in runtime PM resume handler. -- Update dependency list. drivers/soundwire/qcom.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index da1ad7e..445e481 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -13,6 +13,7 @@ #include <linux/of_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/pm_wakeirq.h> #include <linux/slimbus.h> @@ -142,6 +143,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -656,6 +658,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1333,6 +1337,10 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); + ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; @@ -1486,6 +1494,8 @@ static int __maybe_unused swrm_runtime_resume(struct device *dev) qcom_swrm_get_device_status(ctrl); sdw_handle_slave_status(&ctrl->bus, ctrl->status); } else { + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_BUS_CTRL, SWRM_MCP_BUS_CLK_START); ctrl->reg_write(ctrl, SWRM_INTERRUPT_CLEAR, SWRM_INTERRUPT_STATUS_MASTER_CLASH_DET); -- 2.7.4

5 9

[PATCH] ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SSI and SPDIF has dual fifos, enhance P2P for these case with using the sdma_peripheral_config struct Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc_dma.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cb..aaf7993935b7 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -129,6 +129,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *params) { enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + enum sdma_peripheral_type be_peripheral_type = IMX_DMATYPE_SSI; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; @@ -139,6 +140,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; struct dma_slave_config config_fe, config_be; + struct sdma_peripheral_config audio_config; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -221,6 +223,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, /* Get DMA request of Back-End */ tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; + be_peripheral_type = tmp_data->peripheral_type; if (!be_chan) dma_release_channel(tmp_chan); @@ -268,6 +271,17 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, config_be.dst_addr_width = buswidth; config_be.dst_maxburst = dma_params_be->maxburst; + memset(&audio_config, 0, sizeof(audio_config)); + config_be.peripheral_config = &audio_config; + config_be.peripheral_size = sizeof(audio_config); + + if (tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_dst = 2; + if (!tx && (be_peripheral_type == IMX_DMATYPE_SSI_DUAL || + be_peripheral_type == IMX_DMATYPE_SPDIF)) + audio_config.n_fifos_src = 2; + if (tx) { config_be.src_addr = asrc->paddr + asrc->get_fifo_addr(OUT, index); config_be.dst_addr = dma_params_be->addr; -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek - Add HW8326 support
by huangwenhui 07 Jun '22

07 Jun '22

Added the support of new Huawei codec HW8326. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/hda/hdac_device.c | 1 + sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 2 files changed, 15 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..775474f00141 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -651,6 +651,7 @@ static const struct hda_vendor_id hda_vendor_ids[] = { { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, { 0x10ec, "Realtek" }, + { 0x19e5, "Huawei" }, { 0x1102, "Creative" }, { 0x1106, "VIA" }, { 0x111d, "IDT" }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3ad454b3fbf..0f32c0d17a3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -443,6 +443,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -580,6 +581,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -3247,6 +3249,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3275,6 +3278,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -4910,6 +4914,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5025,6 +5030,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -5175,6 +5181,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -5274,6 +5281,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5388,6 +5396,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5489,6 +5498,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5783,6 +5793,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -6385,6 +6396,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); break; @@ -10095,6 +10107,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -11479,6 +11492,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), -- 2.20.1

2 2

[PATCH] ALSA: hda/conexant - Fix loopback issue with CX20632
by huangwenhui 07 Jun '22

07 Jun '22

On a machine with CX20632,Alsamixer doesn't have 'Loopback Mixing' and 'Line'. Signed-off-by: huangwenhui <huangwenhuia(a)uniontech.com> --- sound/pci/hda/patch_conexant.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index aa360a0af284..1248d1a51cf0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1052,6 +1052,13 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f15098: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x22; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + snd_hda_pick_fixup(codec, cxt5066_fixup_models, + cxt5066_fixups, cxt_fixups); + break; case 0x14f150f2: codec->power_save_node = 1; fallthrough; -- 2.20.1

2 1

[PATCH] ALSA: hda: MTL: add HD Audio PCI ID and HDMI codec vendor ID
by Pierre-Louis Bossart 07 Jun '22

07 Jun '22

From: Yong Zhi <yong.zhi(a)intel.com> Add HD Audio PCI ID for Intel Meteorlake platform. Signed-off-by: Kai Vehmanen <kai.vehmanen(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/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a83eb6b88b1f..2c2257ec50929 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2525,6 +2525,9 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cf), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Meteorlake-P */ + { PCI_DEVICE(0x8086, 0x7E28), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 31fe417955712..6c209cd26c0ca 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4554,6 +4554,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281d, "Meteorlake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.34.1

2 1

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… `alsa-info.sh` output: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8846120/… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ALSA: usb-audio: Fix regressions for Saffire 6
by Takashi Iwai 06 Jun '22

06 Jun '22

Hi, this is a small patch set to address the still remaining regressions for Focusrite Saffire 6. Takashi === Takashi Iwai (2): ALSA: usb-audio: Skip generic sync EP parse for secondary EP ALSA: usb-audio: Set up (implicit) sync for Saffire 6 sound/usb/pcm.c | 5 ++++- sound/usb/quirks-table.h | 7 ++++++- 2 files changed, 10 insertions(+), 2 deletions(-) -- 2.35.3

1 2

Add basic support for ALC4082 on ASUS ROG Maximus Z690 Hero
by GitHub pull_request - edited 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf pull request #166 was edited from Lastique: This enables Microphone In in the rear jack and Headphones Out in the front panel jack on ASUS ROG Maximus Z690 Hero with Realtek ALC4082. Things that still don't work: - Microphone In on the front panel - Side channel output on the Line In jack in 7.1 mode - Rear channels and Center/Sub are swapped Line In on the rear panel detects microphone plugged in. Actual input not tested (no line out device available; no input from microphone). S/PDIF not tested. Related to https://github.com/alsa-project/alsa-ucm-conf/issues/165. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add basic support for ALC4082 on ASUS ROG Maximus Z690 Hero
by GitHub pull_request - opened 06 Jun '22

06 Jun '22

alsa-project/alsa-ucm-conf pull request #166 was opened from Lastique: This enables Microphone In in the rear jack and Headphones Out in the front panel jack on ASUS ROG Maximus Z690 Hero with Realtek ALC4082. Things that still don't work: - Microphone In on the front panel - Side channel output on the Line In jack in 7.1 mode - Rear channels and Center/Sub are swapped Line In on the rear panel detects microphone plugged in. Actual input not tested (no line out device available; no input from microphone). S/PDIF not tested. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v3 0/6] OPE support on Tegra210 and later
by Sameer Pujar 06 Jun '22

06 Jun '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v2 -> v3: --------- * Drop "Device Tree Bindings" string from bindind doc titles for OPE, PEQ and MBDRC. v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

1 6

[PATCH v2] ASoC: SOF: Fix potential NULL pointer dereference
by Amadeusz Sławiński 06 Jun '22

06 Jun '22

Cleanup path for sof_prepare_widgets_in_path() should check if unprepare callback exists before calling it, instead it checks if it does not exist. Fix the check. Fixes: 66344c6d9211 ("ASoC: SOF: Add a prepare op to IPC topology widget ops") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- v2 : Fix Fixes hash, apparently I looked at wrong tree --- sound/soc/sof/sof-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 8d740635a4bb..28976098a89e 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -318,7 +318,7 @@ sof_prepare_widgets_in_path(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget p->walking = false; if (ret < 0) { /* unprepare the source widget */ - if (!widget_ops[widget->id].ipc_unprepare && swidget->prepared) { + if (widget_ops[widget->id].ipc_unprepare && swidget->prepared) { widget_ops[widget->id].ipc_unprepare(swidget); swidget->prepared = false; } -- 2.25.1

1 0

[PATCH] slimbus: messaging: fix typos in comments
by Julia Lawall 06 Jun '22

06 Jun '22

Spelling mistakes (triple letters) in comments. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- drivers/slimbus/messaging.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index e5ae26227bdb..4ce0cb61e481 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -79,7 +79,7 @@ int slim_alloc_txn_tid(struct slim_controller *ctrl, struct slim_msg_txn *txn) EXPORT_SYMBOL_GPL(slim_alloc_txn_tid); /** - * slim_free_txn_tid() - Freee tid of txn + * slim_free_txn_tid() - Free tid of txn * * @ctrl: Controller handle * @txn: transaction whose tid should be freed @@ -101,7 +101,7 @@ EXPORT_SYMBOL_GPL(slim_free_txn_tid); * @txn: Transaction to be sent over SLIMbus * * Called by controller to transmit messaging transactions not dealing with - * Interface/Value elements. (e.g. transmittting a message to assign logical + * Interface/Value elements. (e.g. transmitting a message to assign logical * address to a slave device * * Return: -ETIMEDOUT: If transmission of this message timed out

2 1

[PATCH] drivers: slimbus: Directly use ida_alloc()/free()
by keliu 06 Jun '22

06 Jun '22

Use ida_alloc()/ida_free() instead of deprecated ida_simple_get()/ida_simple_remove() . Signed-off-by: keliu <liuke94(a)huawei.com> --- drivers/slimbus/core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 78480e332ab8..219483b79c09 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -250,7 +250,7 @@ int slim_register_controller(struct slim_controller *ctrl) { int id; - id = ida_simple_get(&ctrl_ida, 0, 0, GFP_KERNEL); + id = ida_alloc(&ctrl_ida, GFP_KERNEL); if (id < 0) return id; @@ -299,7 +299,7 @@ int slim_unregister_controller(struct slim_controller *ctrl) { /* Remove all clients */ device_for_each_child(ctrl->dev, NULL, slim_ctrl_remove_device); - ida_simple_remove(&ctrl_ida, ctrl->id); + ida_free(&ctrl_ida, ctrl->id); return 0; } @@ -323,7 +323,7 @@ void slim_report_absent(struct slim_device *sbdev) sbdev->is_laddr_valid = false; mutex_unlock(&ctrl->lock); if (!ctrl->get_laddr) - ida_simple_remove(&ctrl->laddr_ida, sbdev->laddr); + ida_free(&ctrl->laddr_ida, sbdev->laddr); slim_device_update_status(sbdev, SLIM_DEVICE_STATUS_DOWN); } EXPORT_SYMBOL_GPL(slim_report_absent); -- 2.25.1

2 1

[PATCH] ASoC: wm_adsp: Remove redundant NULL check before release_firmware() call
by cgel.zte＠gmail.com 06 Jun '22

06 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> release_firmware() checks for NULL pointers internally so checking before calling it is redundant. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/codecs/wm_adsp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7973a75cac05..4c4344b21c2e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -739,12 +739,10 @@ static void wm_adsp_release_firmware_files(struct wm_adsp *dsp, const struct firmware *coeff_firmware, char *coeff_filename) { - if (wmfw_firmware) - release_firmware(wmfw_firmware); + release_firmware(wmfw_firmware); kfree(wmfw_filename); - if (coeff_firmware) - release_firmware(coeff_firmware); + release_firmware(coeff_firmware); kfree(coeff_filename); } -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Fix for quirk to enable speaker output on the Lenovo Yoga DuetITL 2021
by Cameron Berkenpas 06 Jun '22

06 Jun '22

Enables the ALC287_FIXUP_YOGA7_14ITL_SPEAKERS quirk for the Lenovo Yoga DuetITL 2021 laptop to fix speaker output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> Co-authored-by: Songine <donglingluoying(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 f3ad454b3fbf..2af5725dfca0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9262,6 +9262,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.34.1

2 1

[PATCH] dt-bindings: Fix properties without any type
by Rob Herring 06 Jun '22

06 Jun '22

Now that the schema tools can extract type information for all properties (in order to decode dtb files), finding properties missing any type definition is fairly trivial though not yet automated. Fix the various property schemas which are missing a type. Most of these tend to be device specific properties which don't have a vendor prefix. A vendor prefix is how we normally ensure a type is defined. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../arm/hisilicon/controller/hip04-bootwrapper.yaml | 5 +++-- .../bindings/display/bridge/toshiba,tc358768.yaml | 1 + .../devicetree/bindings/display/panel/panel-timing.yaml | 5 +++++ .../bindings/display/panel/raydium,rm67191.yaml | 1 + .../bindings/display/panel/samsung,s6e8aa0.yaml | 1 + .../devicetree/bindings/gpio/fairchild,74hc595.yaml | 1 + .../devicetree/bindings/input/google,cros-ec-keyb.yaml | 1 + .../devicetree/bindings/input/matrix-keymap.yaml | 4 ++++ Documentation/devicetree/bindings/media/i2c/adv7604.yaml | 3 ++- Documentation/devicetree/bindings/mux/reg-mux.yaml | 8 ++++++-- Documentation/devicetree/bindings/net/cdns,macb.yaml | 1 + Documentation/devicetree/bindings/net/ingenic,mac.yaml | 1 + .../devicetree/bindings/net/ti,davinci-mdio.yaml | 1 + .../devicetree/bindings/net/wireless/ti,wlcore.yaml | 2 ++ .../devicetree/bindings/pci/snps,dw-pcie-ep.yaml | 6 ++++-- Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml | 2 ++ .../devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml | 2 ++ Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml | 1 + .../devicetree/bindings/power/supply/battery.yaml | 7 ++++++- .../devicetree/bindings/power/supply/charger-manager.yaml | 1 + Documentation/devicetree/bindings/rng/st,stm32-rng.yaml | 1 + Documentation/devicetree/bindings/serial/8250.yaml | 1 + .../devicetree/bindings/sound/audio-graph-card2.yaml | 3 +++ .../devicetree/bindings/sound/imx-audio-hdmi.yaml | 3 +++ Documentation/devicetree/bindings/usb/smsc,usb3503.yaml | 1 + 25 files changed, 55 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml b/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml index 7378159e61df..483caf0ce25b 100644 --- a/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml +++ b/Documentation/devicetree/bindings/arm/hisilicon/controller/hip04-bootwrapper.yaml @@ -17,14 +17,15 @@ properties: - const: hisilicon,hip04-bootwrapper boot-method: + $ref: /schemas/types.yaml#/definitions/uint32-array description: | Address and size of boot method. [0]: bootwrapper physical address [1]: bootwrapper size [2]: relocation physical address [3]: relocation size - minItems: 1 - maxItems: 2 + minItems: 2 + maxItems: 4 required: - compatible diff --git a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml index 3bd670b8e5cd..0b6f5bef120f 100644 --- a/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml +++ b/Documentation/devicetree/bindings/display/bridge/toshiba,tc358768.yaml @@ -58,6 +58,7 @@ properties: properties: data-lines: + $ref: /schemas/types.yaml#/definitions/uint32 enum: [ 16, 18, 24 ] port@1: diff --git a/Documentation/devicetree/bindings/display/panel/panel-timing.yaml b/Documentation/devicetree/bindings/display/panel/panel-timing.yaml index 7749de95ee40..229e3b36ee29 100644 --- a/Documentation/devicetree/bindings/display/panel/panel-timing.yaml +++ b/Documentation/devicetree/bindings/display/panel/panel-timing.yaml @@ -146,6 +146,7 @@ properties: Horizontal sync pulse. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] vsync-active: @@ -153,6 +154,7 @@ properties: Vertical sync pulse. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] de-active: @@ -160,6 +162,7 @@ properties: Data enable. 0 selects active low, 1 selects active high. If omitted then it is not used by the hardware + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] pixelclk-active: @@ -169,6 +172,7 @@ properties: sample data on rising edge. Use 1 to drive pixel data on rising edge and sample data on falling edge + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] syncclk-active: @@ -179,6 +183,7 @@ properties: sample sync on rising edge of pixel clock. Use 1 to drive sync on rising edge and sample sync on falling edge of pixel clock + $ref: /schemas/types.yaml#/definitions/uint32 enum: [0, 1] interlaced: diff --git a/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml b/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml index 745dd247c409..617aa8c8c03a 100644 --- a/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml +++ b/Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml @@ -24,6 +24,7 @@ properties: dsi-lanes: description: Number of DSI lanes to be used must be <3> or <4> + $ref: /schemas/types.yaml#/definitions/uint32 enum: [3, 4] v3p3-supply: diff --git a/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml b/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml index ca959451557e..1cdc91b3439f 100644 --- a/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml +++ b/Documentation/devicetree/bindings/display/panel/samsung,s6e8aa0.yaml @@ -36,6 +36,7 @@ properties: init-delay: description: delay after initialization sequence [ms] + $ref: /schemas/types.yaml#/definitions/uint32 panel-width-mm: description: physical panel width [mm] diff --git a/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml b/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml index 5fe19fa5f67c..a99e7842ca17 100644 --- a/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml +++ b/Documentation/devicetree/bindings/gpio/fairchild,74hc595.yaml @@ -26,6 +26,7 @@ properties: const: 2 registers-number: + $ref: /schemas/types.yaml#/definitions/uint32 description: Number of daisy-chained shift registers enable-gpios: diff --git a/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml b/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml index e8f137abb03c..aa61fe64be63 100644 --- a/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml +++ b/Documentation/devicetree/bindings/input/google,cros-ec-keyb.yaml @@ -31,6 +31,7 @@ properties: type: boolean function-row-physmap: + $ref: /schemas/types.yaml#/definitions/uint32-array minItems: 1 maxItems: 15 description: | diff --git a/Documentation/devicetree/bindings/input/matrix-keymap.yaml b/Documentation/devicetree/bindings/input/matrix-keymap.yaml index 6699d5e32dca..9f703bb51e12 100644 --- a/Documentation/devicetree/bindings/input/matrix-keymap.yaml +++ b/Documentation/devicetree/bindings/input/matrix-keymap.yaml @@ -27,6 +27,10 @@ properties: column and linux key-code. The 32-bit big endian cell is packed as: row << 24 | column << 16 | key-code + linux,no-autorepeat: + type: boolean + description: Disable keyrepeat + keypad,num-rows: $ref: /schemas/types.yaml#/definitions/uint32 description: Number of row lines connected to the keypad controller. diff --git a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml index c19d8391e2d5..7589d377c686 100644 --- a/Documentation/devicetree/bindings/media/i2c/adv7604.yaml +++ b/Documentation/devicetree/bindings/media/i2c/adv7604.yaml @@ -60,7 +60,8 @@ properties: enables hot-plug detection. default-input: - maxItems: 1 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 0, 1 ] description: Select which input is selected after reset. diff --git a/Documentation/devicetree/bindings/mux/reg-mux.yaml b/Documentation/devicetree/bindings/mux/reg-mux.yaml index 60d5746eb39d..e2f6b11f1254 100644 --- a/Documentation/devicetree/bindings/mux/reg-mux.yaml +++ b/Documentation/devicetree/bindings/mux/reg-mux.yaml @@ -25,8 +25,12 @@ properties: const: 1 mux-reg-masks: - description: an array of register offset and pre-shifted bitfield mask - pairs, each describing a single mux control. + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: register offset + - description: pre-shifted bitfield mask + description: Each entry describes a single mux control. idle-states: true diff --git a/Documentation/devicetree/bindings/net/cdns,macb.yaml b/Documentation/devicetree/bindings/net/cdns,macb.yaml index 6cd3d853dcba..59fe2789fa44 100644 --- a/Documentation/devicetree/bindings/net/cdns,macb.yaml +++ b/Documentation/devicetree/bindings/net/cdns,macb.yaml @@ -129,6 +129,7 @@ patternProperties: reset-gpios: true magic-packet: + type: boolean description: Indicates that the hardware supports waking up via magic packet. diff --git a/Documentation/devicetree/bindings/net/ingenic,mac.yaml b/Documentation/devicetree/bindings/net/ingenic,mac.yaml index 8e52b2e683b8..93b3e991d209 100644 --- a/Documentation/devicetree/bindings/net/ingenic,mac.yaml +++ b/Documentation/devicetree/bindings/net/ingenic,mac.yaml @@ -37,6 +37,7 @@ properties: const: stmmaceth mode-reg: + $ref: /schemas/types.yaml#/definitions/phandle description: An extra syscon register that control ethernet interface and timing delay rx-clk-delay-ps: diff --git a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml index 6f44f9516c36..a339202c5e8e 100644 --- a/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml +++ b/Documentation/devicetree/bindings/net/ti,davinci-mdio.yaml @@ -34,6 +34,7 @@ properties: maxItems: 1 bus_freq: + $ref: /schemas/types.yaml#/definitions/uint32 maximum: 2500000 description: MDIO Bus frequency diff --git a/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml b/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml index 8dd164d10290..d68bb2ec1f7e 100644 --- a/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml +++ b/Documentation/devicetree/bindings/net/wireless/ti,wlcore.yaml @@ -54,9 +54,11 @@ properties: ref-clock-frequency: + $ref: /schemas/types.yaml#/definitions/uint32 description: Reference clock frequency. tcxo-clock-frequency: + $ref: /schemas/types.yaml#/definitions/uint32 description: TCXO clock frequency. clock-xtal: diff --git a/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml b/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml index e59059ab5be0..b78535040f04 100644 --- a/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml +++ b/Documentation/devicetree/bindings/pci/snps,dw-pcie-ep.yaml @@ -55,13 +55,15 @@ properties: Translation Unit) registers. num-ib-windows: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: number of inbound address translation windows - maxItems: 1 deprecated: true num-ob-windows: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: number of outbound address translation windows - maxItems: 1 deprecated: true required: diff --git a/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml b/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml index a5345c494744..c90e5e2d25f6 100644 --- a/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml +++ b/Documentation/devicetree/bindings/pci/snps,dw-pcie.yaml @@ -68,6 +68,8 @@ properties: Translation Unit) registers. num-viewport: + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 256 description: | number of view ports configured in hardware. If a platform does not specify it, the driver autodetects it. diff --git a/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml b/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml index 53e963e090f2..533b4cfe33d2 100644 --- a/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml +++ b/Documentation/devicetree/bindings/pinctrl/canaan,k210-fpioa.yaml @@ -120,6 +120,7 @@ patternProperties: input-schmitt-disable: true input-polarity-invert: + type: boolean description: Enable or disable pin input polarity inversion. @@ -132,6 +133,7 @@ patternProperties: output-low: true output-polarity-invert: + type: boolean description: Enable or disable pin output polarity inversion. diff --git a/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml b/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml index 3301fa0c2653..301db7daf870 100644 --- a/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml +++ b/Documentation/devicetree/bindings/power/avs/qcom,cpr.yaml @@ -51,6 +51,7 @@ properties: supported by the CPR power domain. acc-syscon: + $ref: /schemas/types.yaml#/definitions/phandle description: A phandle to the syscon used for writing ACC settings. nvmem-cells: diff --git a/Documentation/devicetree/bindings/power/supply/battery.yaml b/Documentation/devicetree/bindings/power/supply/battery.yaml index d56ac484fec5..491488e7b970 100644 --- a/Documentation/devicetree/bindings/power/supply/battery.yaml +++ b/Documentation/devicetree/bindings/power/supply/battery.yaml @@ -85,8 +85,13 @@ properties: description: battery factory internal resistance resistance-temp-table: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + items: + items: + - description: the temperature in degree Celsius + - description: battery internal resistance percent description: | - An array providing the temperature in degree Celsius + A table providing the temperature in degree Celsius and corresponding battery internal resistance percent, which is used to look up the resistance percent according to current temperature to get an accurate batterty internal resistance in different temperatures. diff --git a/Documentation/devicetree/bindings/power/supply/charger-manager.yaml b/Documentation/devicetree/bindings/power/supply/charger-manager.yaml index c863cfa67865..fbb2204769aa 100644 --- a/Documentation/devicetree/bindings/power/supply/charger-manager.yaml +++ b/Documentation/devicetree/bindings/power/supply/charger-manager.yaml @@ -36,6 +36,7 @@ properties: cm-poll-mode: description: polling mode + $ref: /schemas/types.yaml#/definitions/uint32 default: 0 enum: - 0 # disabled diff --git a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml index 9a6e4eaf4d3c..fcd86f822a9c 100644 --- a/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml +++ b/Documentation/devicetree/bindings/rng/st,stm32-rng.yaml @@ -27,6 +27,7 @@ properties: maxItems: 1 clock-error-detect: + type: boolean description: If set enable the clock detection management required: diff --git a/Documentation/devicetree/bindings/serial/8250.yaml b/Documentation/devicetree/bindings/serial/8250.yaml index 3bab2f27b970..5f6b113d378f 100644 --- a/Documentation/devicetree/bindings/serial/8250.yaml +++ b/Documentation/devicetree/bindings/serial/8250.yaml @@ -138,6 +138,7 @@ properties: description: The current active speed of the UART. reg-offset: + $ref: /schemas/types.yaml#/definitions/uint32 description: | Offset to apply to the mapbase from the start of the registers. diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml index f7e94b1e0e4b..7416067c945e 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -24,10 +24,13 @@ properties: connection's sink, the second being the connection's source. $ref: /schemas/types.yaml#/definitions/non-unique-string-array multi: + type: object description: Multi-CPU/Codec node dpcm: + type: object description: DPCM node codec2codec: + type: object description: Codec to Codec node required: diff --git a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml index d5474f83ac2c..e7e7bb65c366 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-hdmi.yaml @@ -20,9 +20,11 @@ properties: description: User specified audio sound card name audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller hdmi-out: + type: boolean description: | This is a boolean property. If present, the transmitting function of HDMI will be enabled, indicating there's a physical HDMI out @@ -30,6 +32,7 @@ properties: block, such as an HDMI encoder or display-controller. hdmi-in: + type: boolean description: | This is a boolean property. If present, the receiving function of HDMI will be enabled, indicating there is a physical HDMI in diff --git a/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml b/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml index b9e219829801..321b6f166197 100644 --- a/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml +++ b/Documentation/devicetree/bindings/usb/smsc,usb3503.yaml @@ -45,6 +45,7 @@ properties: property if all ports have to be enabled. initial-mode: + $ref: /schemas/types.yaml#/definitions/uint32 enum: [1, 2] description: > Specifies initial mode. 1 for Hub mode, 2 for standby mode. -- 2.34.1

8 8

[PATCH] ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare
by cgel.zte＠gmail.com 06 Jun '22

06 Jun '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> Because clk_disable_unprepare already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> --- sound/soc/fsl/imx-audmux.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index dfa05d40b276..f434fa7decc1 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -71,8 +71,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -218,8 +217,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); - if (audmux_clk) - clk_disable_unprepare(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } -- 2.25.1

2 1

HDMI audio broken in 1.2.7 for old laptops
by GitHub issues - opened 06 Jun '22

06 Jun '22

alsa-project/alsa-lib issue #235 was opened from hwpfeil: Tag 1.2.7 breaks HDMI audio on HP laptops with AMD A6-5200 APU (video works). No HDMI option in System Settings|Audio Devices, both KDE and XFCE. Reverting to 1.2.6.1 restores the HDMI Audio Device. There should come a time when such dinosaurs are declared obsolete and no longer supported. They can keep running on 1.2.6.1. Not sure if it needs fixing. Issue URL : https://github.com/alsa-project/alsa-lib/issues/235 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/6] timer: use explicit two types of signals to dispatch events
by GitHub pull_request - opened 05 Jun '22

05 Jun '22

alsa-project/alsa-gobject pull request #74 was opened from takaswie: In current implementation, ALSATimer.UserInstance dispatches one type of event for both tick and timestamp events. However, the type of event is decided before starting timer instance, therefore user application never processes both types of event. In this patchset, two types of signals are added to ALSATimer.UserInstance to dispatch the two types of event separately, then ALSATImer.Event is going to be removed. Additionally, some boxed objects and enumerations are renamed so that they have straight-forward names. ``` Takashi Sakamoto (6): timer: user_instance: use two signals to dispatch events explicitly timer: event: remove unused structure timer: enums: rename TstampEventType from EventType timer: enum: rename EventType from EventDataType timer: tick-event: rename from event-data-tick timer: tstamp-event: rename from event-data-tstamp samples/timer | 29 ++++---- src/timer/alsatimer-enum-types.h | 80 +++++++++++----------- src/timer/alsatimer.h | 5 +- src/timer/alsatimer.map | 30 ++++---- src/timer/event-data-tick.c | 51 -------------- src/timer/event-data-tick.h | 23 ------- src/timer/event-data-tstamp.c | 66 ------------------ src/timer/event-data-tstamp.h | 26 ------- src/timer/event.c | 64 ----------------- src/timer/event.h | 28 -------- src/timer/instance-params.c | 22 +++--- src/timer/instance-params.h | 4 +- src/timer/meson.build | 10 ++- src/timer/tick-event.c | 51 ++++++++++++++ src/timer/tick-event.h | 23 +++++++ src/timer/tstamp-event.c | 64 +++++++++++++++++ src/timer/tstamp-event.h | 24 +++++++ src/timer/user-instance.c | 114 ++++++++++++++++++++----------- src/timer/user-instance.h | 22 ++++-- tests/alsatimer-enums | 6 +- tests/alsatimer-user-instance | 8 ++- 21 files changed, 346 insertions(+), 404 deletions(-) delete mode 100644 src/timer/event-data-tick.c delete mode 100644 src/timer/event-data-tick.h delete mode 100644 src/timer/event-data-tstamp.c delete mode 100644 src/timer/event-data-tstamp.h delete mode 100644 src/timer/event.c delete mode 100644 src/timer/event.h create mode 100644 src/timer/tick-event.c create mode 100644 src/timer/tick-event.h create mode 100644 src/timer/tstamp-event.c create mode 100644 src/timer/tstamp-event.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/74 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/74.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838250/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… This is on Kubuntu 22.04. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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New device naming breaks HDMI audio if ELD lookup fails
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-lib issue #233 was opened from berolinux: The new device naming introduced by commit 859448f01033b40feebf9a0aab467ba57a655b5b breaks HDMI audio if the ELD lookup fails. This happens e.g. with an older Sharp Aquos TV (from 2008) connected to an AMD Radeon HD 7450 graphics card with Linux kernel 5.18.1. Adding some debug statements to control/eld.c shows it fails on the check for `l == 0`, with `l` being `0`, `cinfo.count` being `24`. A hexdump of the 24 bytes in eld is `10 00 05 00 00 10 00 01 00 00 00 00 00 00 00 00 00 00 00 00 09 07 01 00`. The EDID info from the graphics card in sysfs makes sense though: ``` $ edid-decode /sys/class/drm/card0-HDMI-A-1/edid edid-decode (hex): 00 ff ff ff ff ff ff 00 4d 10 22 10 00 00 00 00 ff 12 01 03 80 52 2e 78 2a 1b be a2 55 34 b3 26 14 4a 52 af ce 00 a9 40 90 40 81 80 01 01 01 01 01 01 01 01 01 01 02 3a 80 d0 72 38 2d 40 10 2c 45 80 34 cc 31 00 00 1a 66 21 50 b0 51 00 1b 30 40 70 36 00 00 00 00 00 00 1e 00 00 00 fc 00 53 48 41 52 50 20 48 44 4d 49 0a 20 20 00 00 00 fd 00 17 4c 0f 4b 11 00 0a 20 20 20 20 20 20 01 40 02 03 2e 72 50 9f 90 20 14 05 13 04 12 03 11 02 16 07 15 06 01 23 09 07 01 83 01 00 00 e3 05 03 01 6c 03 0c 00 20 00 80 22 c0 11 11 19 19 02 3a 80 18 71 38 2d 40 58 2c 45 00 34 cc 31 00 00 1e 01 1d 80 d0 72 1c 16 20 10 2c 25 80 34 cc 31 00 00 9e 01 1d 80 18 71 1c 16 20 58 2c 25 00 34 cc 31 00 00 9e 01 1d 00 bc 52 d0 1e 20 b8 28 55 40 34 cc 31 00 00 1e 00 00 00 00 00 00 00 00 00 63 ---------------- Block 0, Base EDID: EDID Structure Version & Revision: 1.3 Vendor & Product Identification: Manufacturer: SHP Model: 4130 Model year: 2008 Basic Display Parameters & Features: Digital display Maximum image size: 82 cm x 46 cm Gamma: 2.20 DPMS levels: Off RGB color display First detailed timing is the preferred timing Color Characteristics: Red : 0.6328, 0.3330 Green: 0.2050, 0.7021 Blue : 0.1503, 0.0810 White: 0.2919, 0.3222 Established Timings I & II: IBM : 720x400 70.082 Hz 9:5 31.467 kHz 28.320 MHz DMT 0x04: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz DMT 0x05: 640x480 72.809 Hz 4:3 37.861 kHz 31.500 MHz DMT 0x06: 640x480 75.000 Hz 4:3 37.500 kHz 31.500 MHz DMT 0x08: 800x600 56.250 Hz 4:3 35.156 kHz 36.000 MHz DMT 0x09: 800x600 60.317 Hz 4:3 37.879 kHz 40.000 MHz DMT 0x0a: 800x600 72.188 Hz 4:3 48.077 kHz 50.000 MHz DMT 0x0b: 800x600 75.000 Hz 4:3 46.875 kHz 49.500 MHz DMT 0x10: 1024x768 60.004 Hz 4:3 48.363 kHz 65.000 MHz DMT 0x11: 1024x768 70.069 Hz 4:3 56.476 kHz 75.000 MHz DMT 0x12: 1024x768 75.029 Hz 4:3 60.023 kHz 78.750 MHz Standard Timings: DMT 0x33: 1600x1200 60.000 Hz 4:3 75.000 kHz 162.000 MHz DMT 0x2a: 1400x1050 59.978 Hz 4:3 65.317 kHz 121.750 MHz DMT 0x23: 1280x1024 60.020 Hz 5:4 63.981 kHz 108.000 MHz Detailed Timing Descriptors: DTD 1: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol N DTD 2: 1360x768 60.015 Hz 85:48 47.712 kHz 85.500 MHz Hfront 64 Hsync 112 Hback 256 Hpol P Vfront 3 Vsync 6 Vback 18 Vpol P Display Product Name: 'SHARP HDMI' Display Range Limits: Monitor ranges (GTF): 23-76 Hz V, 15-75 kHz H, max dotclock 170 MHz Extension blocks: 1 Checksum: 0x40 ---------------- Block 1, CTA-861 Extension Block: Revision: 3 Basic audio support Supports YCbCr 4:4:4 Supports YCbCr 4:2:2 Native detailed modes: 2 Video Data Block: VIC 31: 1920x1080 50.000 Hz 16:9 56.250 kHz 148.500 MHz (native) VIC 16: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (native) VIC 32: 1920x1080 24.000 Hz 16:9 27.000 kHz 74.250 MHz VIC 20: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz VIC 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz VIC 19: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz VIC 4: 1280x720 60.000 Hz 16:9 45.000 kHz 74.250 MHz VIC 18: 720x576 50.000 Hz 16:9 31.250 kHz 27.000 MHz VIC 3: 720x480 59.940 Hz 16:9 31.469 kHz 27.000 MHz VIC 17: 720x576 50.000 Hz 4:3 31.250 kHz 27.000 MHz VIC 2: 720x480 59.940 Hz 4:3 31.469 kHz 27.000 MHz VIC 22: 1440x576i 50.000 Hz 16:9 15.625 kHz 27.000 MHz VIC 7: 1440x480i 59.940 Hz 16:9 15.734 kHz 27.000 MHz VIC 21: 1440x576i 50.000 Hz 4:3 15.625 kHz 27.000 MHz VIC 6: 1440x480i 59.940 Hz 4:3 15.734 kHz 27.000 MHz VIC 1: 640x480 59.940 Hz 4:3 31.469 kHz 25.175 MHz Audio Data Block: Linear PCM: Max channels: 2 Supported sample rates (kHz): 48 44.1 32 Supported sample sizes (bits): 16 Speaker Allocation Data Block: FL/FR - Front Left/Right Colorimetry Data Block: xvYCC601 xvYCC709 Vendor-Specific Data Block (HDMI), OUI 00-0C-03: Source physical address: 2.0.0.0 Supports_AI Maximum TMDS clock: 170 MHz Video latency: 35 ms Audio latency: 35 ms Interlaced video latency: 51 ms Interlaced audio latency: 51 ms Detailed Timing Descriptors: DTD 3: 1920x1080 60.000 Hz 16:9 67.500 kHz 148.500 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 4 Vsync 5 Vback 36 Vpol P DTD 4: 1920x1080i 50.000 Hz 16:9 28.125 kHz 74.250 MHz (820 mm x 460 mm) Hfront 528 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 5: 1920x1080i 60.000 Hz 16:9 33.750 kHz 74.250 MHz (820 mm x 460 mm) Hfront 88 Hsync 44 Hback 148 Hpol P Vfront 2 Vsync 5 Vback 15 Vpol P Vfront +0.5 Odd Field Vfront 2 Vsync 5 Vback 15 Vpol P Vback +0.5 Even Field DTD 6: 1280x720 50.000 Hz 16:9 37.500 kHz 74.250 MHz (820 mm x 460 mm) Hfront 440 Hsync 40 Hback 220 Hpol P Vfront 5 Vsync 5 Vback 20 Vpol P Checksum: 0x63 ``` Since an error code (`-EIO`) is passed on from `__snd_pcm_info_eld_fixup`, the HDMI audio output goes away altogether instead of just keeping the old name. Will send a PR to keep the audio working, but it would be nice to actually make the new naming work as well. Issue URL : https://github.com/alsa-project/alsa-lib/issues/233 Repository URL: https://github.com/alsa-project/alsa-lib

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ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - edited 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was edited from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. I also tried to connect the headset to the front panel connectors, but got no audio output nor capture. This is not my intended configuration though, I normally connect the headset to the rear panel. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus Z690 Hero (ALC4082) - No audio from microphone, no audio on the front panel
by GitHub issues - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf issue #165 was opened from Lastique: I'm having a problem with getting audio from the microphone connected to the rear panel jack. Also, if I connect the headset to the front panel headphone/mic jacks, I get neither audio out or in. My main concern is the rear panel jack though. The motherboard is ASUS ROG Maximus Z690 Hero, ALC4082 USB Audio chipset. I have tried this: 1. Installed linux kernel 5.18.0. 2. Copied `ucm2` from alsa-ucm-conf master (rev ef3b6513b6581705d36432428a7f6a4ae23b1276 which is after https://github.com/alsa-project/alsa-ucm-conf/pull/143 was merged) to `/usr/share/alsa`. 3. In `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` added hardware id of my USB Audio device (`0b05:1a27`) to the `realtek-alc4080` rule. Also added the hardware id in `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf`, in the `asus-rog-usb` block condition. 4. Rebooted. 5. The headset is connected to the rear panel (Front out and Mic in jacks; also tried Line in). There are three devices for the USB Audio card, and there are three volume controls in alsamixer: "Line", "Mic" and "Analog In". The problem is that there is no audio coming from the first two devices (which are presumably "Line" and "Mic"). I can record audio from the third device ("Analog In") when the microphone is connected to Mic jack. Unfortunately, "Analog In" cannot be selected in PulseAudio for some reason (I created a [bug](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1364) for PA), which means no audio can be recorded. `arecord` output: [arecord.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838239/ar… `lsusb` output: [lsusb.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8838241/lsus… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/165 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Fixes for veyron chromebooks
by GitHub pull_request - opened 04 Jun '22

04 Jun '22

alsa-project/alsa-ucm-conf pull request #164 was opened from jenneron: Add PlaybackVolume and JackControl. I didn't add JackControl for headset mic because internal mic breaks after switching to headset and back Add symlink to VEYRON-I2S, so ucm config will get probed Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/164.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Re: [Sound-open-firmware] jack detection via input/event on linux
by Pierre-Louis Bossart 03 Jun '22

03 Jun '22

[Adding alsa-devel, Mark and Takashi to this thread] On 6/3/22 08:33, Jaroslav Kysela wrote: > On 03. 06. 22 13:17, hede wrote: >> Am 03.06.2022 11:34, schrieb Jaroslav Kysela: >>> PA/PW use only ALSA controls at the moment - thus you should define >>> JackControl as defined in use-case.h. But it seems that the driver >>> does not create or publish the ALSA jack control (I see only >>> "iface=CARD,name='HDMI/DP,pcm=2 Jack'" jack controls for HDMI). >>> >>> Jaroslav >> >> Ah, thank you Jaroslav. That's the hint I was searching for atm. >> JackDev vs. JackControl. I had a vague guess on this but now you made it >> certain. >> >> Maybe I should take a look into the driver directly. > > It seems that many ASoC drivers do not create kctls - the argument pins > is zero in the snd_soc_card_jack_new() call. The SND_JACK_HEADSET should > be remapped to kctl in sound/soc/intel/boards/sof_rt5682.c, too. Did you mean something like the patch below Jaroslav? If yes, most of the Chromebook machine drivers are missing this remapping. That would also explain why the jack detection didn't work on my GLK test device, despite a UCM file that looked good... diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5d67a2c87a1d4..1db17ebd26f63 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -85,6 +85,17 @@ struct sof_card_private { bool idisp_codec; }; +static struct snd_soc_jack_pin sof_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) { sof_rt5682_quirk = (unsigned long)id->driver_data; @@ -295,11 +306,13 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) * Headset buttons map to the google Reference headset. * These can be configured by userspace. */ - ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->sof_headset); + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, + sof_jack_pins, + ARRAY_SIZE(sof_jack_pins)); if (ret) { dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); return ret; @@ -481,7 +494,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) } static const struct snd_kcontrol_new sof_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), @@ -489,7 +502,7 @@ static const struct snd_kcontrol_new sof_controls[] = { }; static const struct snd_soc_dapm_widget sof_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Left Spk", NULL), SND_SOC_DAPM_SPK("Right Spk", NULL), @@ -501,8 +514,8 @@ static const struct snd_soc_dapm_widget dmic_widgets[] = { static const struct snd_soc_dapm_route sof_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, + { "Headphone", NULL, "HPOL" }, + { "Headphone", NULL, "HPOR" }, /* other jacks */ { "IN1P", NULL, "Headset Mic" },

3 3

[PATCH] ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe
by Miaoqian Lin 03 Jun '22

03 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Add missing of_node_put() to avoid refcount leak. Fixes: f0ab0bf250da ("ASoC: add mt6797-mt6351 driver and config option") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- changes in v2: - update Fixes tag v1 link: https://lore.kernel.org/r/20220603083417.9011-1-linmq006@gmail.com/ --- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index 496f32bcfb5e..d2f6213a6bfc 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -217,7 +217,8 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -230,6 +231,9 @@ static int mt6797_mt6351_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.25.1

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[PATCH 0/6] Fixup Cirrus SX control usage
by Charles Keepax 03 Jun '22

03 Jun '22

Mostly the usage of the SX controls seems to match the lowest gain value + number of gain levels expected. The one notable exception there being cs53l30 as David noted. However, there are a couple of other places where the minimum value/TLVs are slightly incorrectly specified. There are two I couldn't quite work out yet, any input on these greatly appreciated: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), 0.5dB step 0x40 - -96dB 0x00 - 0dB This one is weird, the datasheet says 0x40 is the minimum, which matches, but the datasheet also says 0.5dB steps to 0x00 being 0dB which doesn't match with 0x40 being the minimum, or 0xC0 being the number of levels. Some clarification will needed from the hardware guys. SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, 0, 0x07, 0x23, beep_tlv), 2dB step 0x07 - -50dB 0x00 - 0dB 0x06 - +6dB The minimum checks out at 0x07, but rest of this is weird. The values in the datasheet don't make sense, based on starting at -50dB with a 2dB step, 0x06 should equate to 12dB, over 31 volume levels, not 6dB over 35 volume levels as stated. The control specifies 0x23(35) levels, which takes us up to a final value of 0x0A at 20dB. Again some clarification from the hardware guys might be needed here. Finally it is worth noting I don't actually have hardware for most of these devices, so mostly this is just build tested. Thanks, Charles Charles Keepax (6): ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control sound/soc/codecs/cs35l36.c | 3 ++- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 8 +++++--- sound/soc/codecs/cs42l56.c | 4 ++-- sound/soc/codecs/cs53l30.c | 16 ++++++++-------- 5 files changed, 18 insertions(+), 15 deletions(-) -- 2.30.2

2 7

[PATCH v2] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Brent Lu 03 Jun '22

03 Jun '22

From: xliu <xiang.liu(a)cirrus.com> The default mapping of ASPRX1 (DAC source) is slot 0. Change the slot mapping of right amplifiers (WR and TR) to slot 1 to receive right channel data. Also update the ACPI instance ID mapping according to HW configuration. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 40 +++++++++++++++++++--- 1 file changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..f4192df962d6 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -54,22 +54,29 @@ static struct snd_soc_dai_link_component cs35l41_components[] = { }, }; +/* + * Mapping between ACPI instance id and speaker position. + * + * Four speakers: + * 0: Tweeter left, 1: Woofer left + * 2: Tweeter right, 3: Woofer right + */ static struct snd_soc_codec_conf cs35l41_codec_conf[] = { { .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "WL", + .name_prefix = "TL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WR", + .name_prefix = "WL", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TL", + .name_prefix = "TR", }, { .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "TR", + .name_prefix = "WR", }, }; @@ -101,6 +108,21 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) return ret; } +/* + * Channel map: + * + * TL/WL: ASPRX1 on slot 0, ASPRX2 on slot 1 (default) + * TR/WR: ASPRX1 on slot 1, ASPRX2 on slot 0 + */ +static const struct { + unsigned int rx[2]; +} cs35l41_channel_map[] = { + {.rx = {0, 1}}, /* TL */ + {.rx = {0, 1}}, /* WL */ + {.rx = {1, 0}}, /* TR */ + {.rx = {1, 0}}, /* WR */ +}; + static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -134,6 +156,16 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* setup channel map */ + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(cs35l41_channel_map[i].rx), + (unsigned int *)cs35l41_channel_map[i].rx); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } } return 0; -- 2.25.1

4 3

[PATCH V0 1/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 03 Jun '22

03 Jun '22

use hashtable instead of linear list to check kcontrol before adding them for improving early audio KPI. With this changes we see 600ms improvement in start of audio Change-Id: I1a95d88ef41b48430b57307a0e6a8e82788b4372 Signed-off-by: Raghu Bankapur <quic_rbankapu(a)quicinc.com> --- include/sound/control.h | 2 + include/sound/core.h | 5 +- sound/core/control.c | 106 +++++++++++++++++++++++----------------- sound/core/init.c | 1 + 4 files changed, 68 insertions(+), 46 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..e50db5c45114 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -70,6 +70,8 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; + struct hlist_node hnode; + unsigned int knametoint; /* kctl name to uint, hash key value */ unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..90341d6f1573 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/hashtable.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -24,6 +25,8 @@ #define CONFIG_SND_MAJOR 116 /* standard configuration */ +#define SND_CTL_HASH_TABLE_BITS 14 /* buckets numbers: 1 << 14 */ + /* forward declarations */ struct pci_dev; struct module; @@ -103,7 +106,7 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ - + DECLARE_HASHTABLE(ctl_htable, SND_CTL_HASH_TABLE_BITS); struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..e00a02015837 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -331,43 +331,49 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) } EXPORT_SYMBOL(snd_ctl_free_one); -static bool snd_ctl_remove_numid_conflict(struct snd_card *card, - unsigned int count) +enum snd_ctl_add_mode { + CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, +}; + +char snd_ctl_string[50] = { '\0' }; + +/* Used to convert the string into int value -- BKDRHash */ +static unsigned int snd_ctl_strtoint(const char *s) { - struct snd_kcontrol *kctl; + unsigned int res = 0; - /* Make sure that the ids assigned to the control do not wrap around */ - if (card->last_numid >= UINT_MAX - count) - card->last_numid = 0; + while (*s) + res = (res << 5) - res + (*s++); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid < card->last_numid + 1 + count && - kctl->id.numid + kctl->count > card->last_numid + 1) { - card->last_numid = kctl->id.numid + kctl->count - 1; - return true; - } - } - return false; + return (res & 0x7FFFFFFF); } -static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) +/** + * snd_ctl_hash_check - Check the duplicate enrty on snd hashtable + * @card: the card instance + * @nametoint: kctl name to uint + * + * Finds the control instance with the given nametoint from the card. + * + * Return: The pointer of the instance if found, or %NULL if not. + * + */ +static struct snd_kcontrol *snd_ctl_hash_check(struct snd_card *card, + unsigned int nametoint) { - unsigned int iter = 100000; + struct snd_kcontrol *kctl = NULL; - while (snd_ctl_remove_numid_conflict(card, count)) { - if (--iter == 0) { - /* this situation is very unlikely */ - dev_err(card->dev, "unable to allocate new control numid\n"); - return -ENOMEM; - } + if (snd_BUG_ON(!card)) + return NULL; + + hash_for_each_possible(card->ctl_htable, kctl, hnode, nametoint) { + if (kctl->knametoint != nametoint) + continue; + return kctl; } - return 0; + return NULL; } -enum snd_ctl_add_mode { - CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, -}; - /* add/replace a new kcontrol object; call with card->controls_rwsem locked */ static int __snd_ctl_add_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, @@ -382,26 +388,34 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (id.index > UINT_MAX - kcontrol->count) return -EINVAL; - old = snd_ctl_find_id(card, &id); - if (!old) { - if (mode == CTL_REPLACE) - return -EINVAL; - } else { - if (mode == CTL_ADD_EXCLUSIVE) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i is already present\n", - id.iface, id.device, id.subdevice, id.name, - id.index); - return -EBUSY; - } + snprintf(snd_ctl_string, strlen(kcontrol->id.name) + 6, "%s%d%d%d", + kcontrol->id.name, kcontrol->id.iface, kcontrol->id.device, + kcontrol->id.subdevice); - err = snd_ctl_remove(card, old); - if (err < 0) - return err; - } + kcontrol->knametoint = snd_ctl_strtoint(snd_ctl_string); + if (kcontrol->knametoint < 0) + return -EINVAL; - if (snd_ctl_find_hole(card, kcontrol->count) < 0) - return -ENOMEM; + old = snd_ctl_hash_check(card, kcontrol->knametoint); + if (old) { + old = snd_ctl_find_id(card, &id); + if (!old) { + if (mode == CTL_REPLACE) + return -EINVAL; + } else { + if (mode == CTL_ADD_EXCLUSIVE) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, + id.index); + return -EBUSY; + } + + err = snd_ctl_remove(card, old); + if (err < 0) + return err; + } + } list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; @@ -411,6 +425,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); + hash_add(card->ctl_htable, &kcontrol->hnode, kcontrol->knametoint); + return 0; } diff --git a/sound/core/init.c b/sound/core/init.c index 31ba7024e3ad..24138902e5f2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -284,6 +284,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); + hash_init(card->ctl_htable); mutex_init(&card->memory_mutex); #ifdef CONFIG_PM init_waitqueue_head(&card->power_sleep); -- 2.17.1

1 0

[PATCH V1 1/1] asoc: msm: use hashtable to check kcontrol
by Raghu Bankapur 03 Jun '22

03 Jun '22

use hashtabe instead of linear list to check kcontrol before adding them for improving early audio KPI. Change-Id: I7134816736e08e338c0f22a8ae283a0520aa847a Signed-off-by: Raghu Bankapur <quic_rbankapu(a)quicinc.com> --- include/sound/control.h | 2 + include/sound/core.h | 5 +- sound/core/control.c | 106 +++++++++++++++++++++++----------------- sound/core/init.c | 1 + 4 files changed, 68 insertions(+), 46 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..e50db5c45114 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -70,6 +70,8 @@ struct snd_kcontrol_volatile { struct snd_kcontrol { struct list_head list; /* list of controls */ struct snd_ctl_elem_id id; + struct hlist_node hnode; + unsigned int knametoint; /* kctl name to uint, hash key value */ unsigned int count; /* count of same elements */ snd_kcontrol_info_t *info; snd_kcontrol_get_t *get; diff --git a/include/sound/core.h b/include/sound/core.h index b7e9b58d3c78..90341d6f1573 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -14,6 +14,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/stringify.h> #include <linux/printk.h> +#include <linux/hashtable.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -24,6 +25,8 @@ #define CONFIG_SND_MAJOR 116 /* standard configuration */ +#define SND_CTL_HASH_TABLE_BITS 14 /* buckets numbers: 1 << 14 */ + /* forward declarations */ struct pci_dev; struct module; @@ -103,7 +106,7 @@ struct snd_card { size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ - + DECLARE_HASHTABLE(ctl_htable, SND_CTL_HASH_TABLE_BITS); struct snd_info_entry *proc_root; /* root for soundcard specific files */ struct proc_dir_entry *proc_root_link; /* number link to real id */ diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..e00a02015837 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -331,43 +331,49 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) } EXPORT_SYMBOL(snd_ctl_free_one); -static bool snd_ctl_remove_numid_conflict(struct snd_card *card, - unsigned int count) +enum snd_ctl_add_mode { + CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, +}; + +char snd_ctl_string[50] = { '\0' }; + +/* Used to convert the string into int value -- BKDRHash */ +static unsigned int snd_ctl_strtoint(const char *s) { - struct snd_kcontrol *kctl; + unsigned int res = 0; - /* Make sure that the ids assigned to the control do not wrap around */ - if (card->last_numid >= UINT_MAX - count) - card->last_numid = 0; + while (*s) + res = (res << 5) - res + (*s++); - list_for_each_entry(kctl, &card->controls, list) { - if (kctl->id.numid < card->last_numid + 1 + count && - kctl->id.numid + kctl->count > card->last_numid + 1) { - card->last_numid = kctl->id.numid + kctl->count - 1; - return true; - } - } - return false; + return (res & 0x7FFFFFFF); } -static int snd_ctl_find_hole(struct snd_card *card, unsigned int count) +/** + * snd_ctl_hash_check - Check the duplicate enrty on snd hashtable + * @card: the card instance + * @nametoint: kctl name to uint + * + * Finds the control instance with the given nametoint from the card. + * + * Return: The pointer of the instance if found, or %NULL if not. + * + */ +static struct snd_kcontrol *snd_ctl_hash_check(struct snd_card *card, + unsigned int nametoint) { - unsigned int iter = 100000; + struct snd_kcontrol *kctl = NULL; - while (snd_ctl_remove_numid_conflict(card, count)) { - if (--iter == 0) { - /* this situation is very unlikely */ - dev_err(card->dev, "unable to allocate new control numid\n"); - return -ENOMEM; - } + if (snd_BUG_ON(!card)) + return NULL; + + hash_for_each_possible(card->ctl_htable, kctl, hnode, nametoint) { + if (kctl->knametoint != nametoint) + continue; + return kctl; } - return 0; + return NULL; } -enum snd_ctl_add_mode { - CTL_ADD_EXCLUSIVE, CTL_REPLACE, CTL_ADD_ON_REPLACE, -}; - /* add/replace a new kcontrol object; call with card->controls_rwsem locked */ static int __snd_ctl_add_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, @@ -382,26 +388,34 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (id.index > UINT_MAX - kcontrol->count) return -EINVAL; - old = snd_ctl_find_id(card, &id); - if (!old) { - if (mode == CTL_REPLACE) - return -EINVAL; - } else { - if (mode == CTL_ADD_EXCLUSIVE) { - dev_err(card->dev, - "control %i:%i:%i:%s:%i is already present\n", - id.iface, id.device, id.subdevice, id.name, - id.index); - return -EBUSY; - } + snprintf(snd_ctl_string, strlen(kcontrol->id.name) + 6, "%s%d%d%d", + kcontrol->id.name, kcontrol->id.iface, kcontrol->id.device, + kcontrol->id.subdevice); - err = snd_ctl_remove(card, old); - if (err < 0) - return err; - } + kcontrol->knametoint = snd_ctl_strtoint(snd_ctl_string); + if (kcontrol->knametoint < 0) + return -EINVAL; - if (snd_ctl_find_hole(card, kcontrol->count) < 0) - return -ENOMEM; + old = snd_ctl_hash_check(card, kcontrol->knametoint); + if (old) { + old = snd_ctl_find_id(card, &id); + if (!old) { + if (mode == CTL_REPLACE) + return -EINVAL; + } else { + if (mode == CTL_ADD_EXCLUSIVE) { + dev_err(card->dev, + "control %i:%i:%i:%s:%i is already present\n", + id.iface, id.device, id.subdevice, id.name, + id.index); + return -EBUSY; + } + + err = snd_ctl_remove(card, old); + if (err < 0) + return err; + } + } list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; @@ -411,6 +425,8 @@ static int __snd_ctl_add_replace(struct snd_card *card, for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_ADD, kcontrol, idx); + hash_add(card->ctl_htable, &kcontrol->hnode, kcontrol->knametoint); + return 0; } diff --git a/sound/core/init.c b/sound/core/init.c index 31ba7024e3ad..24138902e5f2 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -284,6 +284,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); INIT_LIST_HEAD(&card->files_list); + hash_init(card->ctl_htable); mutex_init(&card->memory_mutex); #ifdef CONFIG_PM init_waitqueue_head(&card->power_sleep); -- 2.17.1

2 1

[PATCH] ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe
by Miaoqian Lin 03 Jun '22

03 Jun '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when not need anymore. Fix refcount leak in some error paths. Fixes: format:0f83f9296d5c ("ASoC: mediatek: Add machine driver for ALC5650 codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index d1c94acb4516..e05f2b0231fe 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -280,7 +280,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].of_node = mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[0].of_node; @@ -293,7 +294,7 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", __func__, ret); - return ret; + goto put_platform_node; } mt8173_rt5650_dais[DAI_LINK_CODEC_I2S].codecs[1].dai_name = codec_capture_dai; @@ -315,12 +316,14 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (!mt8173_rt5650_dais[DAI_LINK_HDMI_I2S].codecs->of_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); +put_platform_node: of_node_put(platform_node); return ret; } -- 2.25.1

1 0

[bug report] ASoC: SOF: ipc-msg-injector: Separate the message sending
by Dan Carpenter 03 Jun '22

03 Jun '22

Hello Peter Ujfalusi, The patch a9aa3381e404: "ASoC: SOF: ipc-msg-injector: Separate the message sending" from May 6, 2022, leads to the following Smatch static checker warning: sound/soc/sof/sof-client-ipc-msg-injector.c:162 sof_msg_inject_dfs_write() warn: kernel error codes cast to unsigned 'size' sound/soc/sof/sof-client-ipc-msg-injector.c 148 static ssize_t sof_msg_inject_dfs_write(struct file *file, const char __user *buffer, 149 size_t count, loff_t *ppos) 150 { 151 struct sof_client_dev *cdev = file->private_data; 152 struct sof_msg_inject_priv *priv = cdev->data; 153 size_t size; 154 int ret; 155 156 if (*ppos) 157 return 0; I think there needs to be an "if (count != priv->max_msg_size)" check or something. Or another option would be the do a memset() memset(priv->tx_buffer, 0, priv->max_msg_size); before the simple_write_to_buffer(). Otherwise if count == 1 then we re-use stale data. 158 159 size = simple_write_to_buffer(priv->tx_buffer, priv->max_msg_size, 160 ppos, buffer, count); 161 if (size != count) --> 162 return size > 0 ? -EFAULT : size; 163 164 memset(priv->rx_buffer, 0, priv->max_msg_size); 165 166 ret = sof_msg_inject_send_message(cdev); 167 168 /* return the error code if test failed */ 169 if (ret < 0) 170 size = ret; 171 172 return size; 173 }; regards, dan carpenter

2 1

Issues with ALSA after update to Pop!_OS 22.04 LTS Jaunty Pipewire
by Mikhail Kravchenko 03 Jun '22

03 Jun '22

I updated the operating system running on my machine to 22.04 Jaunty. This update brings the system over to pipe-wire. After running certain programs that utilize alsa, the audio on my system becomes intermittent and runs only one application's audio at a time. Resetting the pipe-wire services and forcing alsa to reload its modules does not reliably mitigate the issue. This is the url generated by alsa-info. http://alsa-project.org/db/?f=4d643a3153ffabd5c72313e8ffa3c9f393e676ef On a power cycle, the audio seems to work fine, but as soon as I run the problematic program I get the output: ALSA lib pcm_dmix.c:1032:(snd_pcm_dmix_open) unable to open slave If the program's audio is successful, and it opens the slave. Audio to other applications is upset. sudo alsa force-reload Unloading ALSA sound driver modules: snd-seq-dummy snd-seq-midi snd-seq-midi-event snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer (failed: modules still loaded: snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer). Loading ALSA sound driver modules: snd-seq-dummy snd-seq-midi snd-seq-midi-event snd-hda-codec-realtek snd-hda-codec-generic snd-hda-codec-hdmi snd-hda-intel snd-intel-dspcfg snd-intel-sdw-acpi snd-hda-codec snd-hda-core snd-hrtimer snd-usb-audio snd-usbmidi-lib snd-hwdep snd-rawmidi snd-seq snd-pcm snd-seq-device snd-timer. MK

1 0

USB-Audio: Steinberg UR44 not getting expected results
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-ucm-conf issue #163 was opened from CameronNemo: I am testing the following software: * alsa-lib: 1.2.7 * alsa-utils: 1.2.7 * alsa-ucm-conf: 1.2.7 Unfortunately the changes for the Steinberg UR44 do not seem to be working as intended. I am not seeing the expected mono inputs and stereo outputs that I see defined in `/usr/share/alsa/ucm2/USB-Audio/Steinberg/UR44-HiFi.conf`. I am seeing the following from `aplay -L`: ``` null Discard all samples (playback) or generate zero samples (capture) default Default Audio Device sysdefault Default Audio Device default:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device sysdefault:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device front:CARD=UR44,DEV=0 Steinberg UR44, USB Audio Front output / input surround21:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=UR44,DEV=0 Steinberg UR44, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers ``` And the following from `arecord -L`: ``` null Discard all samples (playback) or generate zero samples (capture) default Default Audio Device sysdefault Default Audio Device default:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device sysdefault:CARD=UR44 Steinberg UR44, USB Audio Default Audio Device front:CARD=UR44,DEV=0 Steinberg UR44, USB Audio Front output / input ``` `spa-acp-tool list`: ``` card 0: profiles:2 devices:2 ports:2 profile 0: name:"off" prio:0 (available: yes) * profile 1: name:"Direct" prio:1 (available: unknown) * port 0: name:"[Out] Direct" direction:playback prio:1000 (available: unknown) * port 1: name:"[In] Direct" direction:capture prio:1000 (available: unknown) * device 0: direction:playback name:"Direct: hw:UR44: sink" prio:1000 flags:00000009 devices: "hw:UR44" * device 1: direction:capture name:"Direct: hw:UR44: source" prio:1000 flags:00000009 devices: "hw:UR44" ``` `pactl list sources`: ``` Source #56 State: SUSPENDED Name: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink.monitor Description: Monitor of Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 4ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right Owner Module: 4294967295 Mute: no Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB, rear-left: 65536 / 100% / 0.00 dB, rear-right: 65536 / 100% / 0.00 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor of Sink: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:playback" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "playback" audio.channels = "4" audio.position = "FL,FR,RL,RR" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "monitor" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: sink" device.profile.description = "Direct Steinberg UR44" card.profile.device = "0" device.id = "54" factory.name = "api.alsa.pcm.sink" priority.driver = "1000" priority.session = "1000" media.class = "Audio/Sink" node.nick = "Steinberg UR44" node.name = "alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-card" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "merge" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "56" object.serial = "56" Formats: pcm Source #57 State: SUSPENDED Name: alsa_input.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__source Description: Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 6ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right,front-center,lfe Owner Module: 4294967295 Mute: no Volume: front-left: 48287 / 74% / -7.96 dB, front-right: 48287 / 74% / -7.96 dB, rear-left: 48287 / 74% / -7.96 dB, rear-right: 48287 / 74% / -7.96 dB, front-center: 48287 / 74% / -7.96 dB, lfe: 48287 / 74% / -7.96 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor of Sink: n/a Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:capture" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "capture" audio.channels = "6" audio.position = "FL,FR,RL,RR,FC,LFE" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: source" device.profile.description = "Direct Steinberg UR44" card.profile.device = "1" device.id = "54" factory.name = "api.alsa.pcm.source" priority.driver = "2000" priority.session = "2000" media.class = "Audio/Source" node.nick = "Steinberg UR44" node.name = "alsa_input.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__source" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-input-microphone" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "split" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "57" object.serial = "57" Ports: [In] Direct: Direct Steinberg UR44 (type: Unknown, priority: 1000, availability unknown) Active Port: [In] Direct Formats: pcm ``` `pactl list sinks`: ``` Sink #56 State: SUSPENDED Name: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink Description: Steinberg UR44 Direct Steinberg UR44 Driver: PipeWire Sample Specification: s24le 4ch 48000Hz Channel Map: front-left,front-right,rear-left,rear-right Owner Module: 4294967295 Mute: no Volume: front-left: 48287 / 74% / -7.96 dB, front-right: 48287 / 74% / -7.96 dB, rear-left: 48287 / 74% / -7.96 dB, rear-right: 48287 / 74% / -7.96 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor Source: alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink.monitor Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: object.path = "alsa:pcm:0:hw:UR44:playback" api.alsa.path = "hw:UR44" api.alsa.open.ucm = "true" api.alsa.pcm.card = "0" api.alsa.pcm.stream = "playback" audio.channels = "4" audio.position = "FL,FR,RL,RR" device.routes = "1" alsa.mixer_device = "_ucm0003.hw:UR44" alsa.resolution_bits = "24" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "USB Audio" alsa.id = "USB Audio" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "Steinberg UR44" alsa.long_card_name = "Yamaha Corporation Steinberg UR44 at usb-0000:04:00.4-1, high speed" alsa.driver_name = "snd_usb_audio" device.profile.name = "Direct: hw:UR44: sink" device.profile.description = "Direct Steinberg UR44" card.profile.device = "0" device.id = "54" factory.name = "api.alsa.pcm.sink" priority.driver = "1000" priority.session = "1000" media.class = "Audio/Sink" node.nick = "Steinberg UR44" node.name = "alsa_output.usb-Yamaha_Corporation_Steinberg_UR44-00.Direct__hw_UR44__sink" device.description = "Steinberg UR44 Direct Steinberg UR44" device.icon_name = "audio-card" device.bus = "usb" device.bus_path = "pci-0000:04:00.4-usb-0:1:1.0" node.pause-on-idle = "false" factory.id = "18" client.id = "34" clock.quantum-limit = "8192" node.driver = "true" factory.mode = "merge" audio.adapt.follower = "" library.name = "audioconvert/libspa-audioconvert" object.id = "56" object.serial = "56" Ports: [Out] Direct: Direct Steinberg UR44 (type: Unknown, priority: 1000, availability unknown) Active Port: [Out] Direct Formats: pcm ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/163 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/4] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (4): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface hwdep: device-info: allow to set property at instantiation src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/device-info.c | 38 ++++++- src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 8 files changed, 389 insertions(+), 46 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 0/4] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 0/3] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - edited 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was edited from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (3): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device hwdep: add error domain for implementators of DeviceCommon interface src/hwdep/alsahwdep-enum-types.h | 89 +++++++++------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 10 ++ src/hwdep/device-common.c | 178 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 106 ++++++++++++++++++ src/hwdep/meson.build | 2 + tests/alsahwdep-enums | 10 ++ 7 files changed, 356 insertions(+), 41 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] hwdep: define interface for hwdep device
by GitHub pull_request - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-gobject pull request #73 was opened from takaswie: The protocol in ALSA HwDep character device has common feature to each device. It's suitable for userspace implementation to have GObject interface for such features. This patchset adds the interface. The implementation of interface should have four methods; open, get_protocol_version, get_device_info, and create_source. Additionally, it should emit handle-disconnection signal when detecting ENODEV error in the interation of source. It requires to notify userspace application since the application should release character device immediately. ``` Takashi Sakamoto (2): hwdep: use explicit value for iface type enumeration hwdep: device-common: add interface for common feature of ALSA HwDep device src/hwdep/alsahwdep-enum-types.h | 67 +++++++--------- src/hwdep/alsahwdep.h | 2 + src/hwdep/alsahwdep.map | 6 ++ src/hwdep/device-common.c | 128 +++++++++++++++++++++++++++++++ src/hwdep/device-common.h | 102 ++++++++++++++++++++++++ src/hwdep/meson.build | 2 + 6 files changed, 267 insertions(+), 40 deletions(-) create mode 100644 src/hwdep/device-common.c create mode 100644 src/hwdep/device-common.h ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/73 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Semantics for _SX controls
by Mark Brown 02 Jun '22

02 Jun '22

Hi, The fixes for input validation have highlighed a bunch of problems with the _SX controls. These aren't widely used, it's just some Cirrus Logic and Qualcomm CODECs plus the obsolte ab8500 driver, but it has become apparent that at least some of the drivers were relying on the lack of validation we used to have to work. For the Cirrus case 34198710f55b5f ("ASoC: Add info callback for SX_TLV controls") says that the intended semantic is Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. which was from Charles at Cirrus so hopefully that's the semantic used by Cirrus drivers. However Tan Nayir pointed at some out of tree patches from Qualcomm which change the info callback such that max is instead the maximum register value that can be written and reports that since we started validating written values properly they can no longer set some valid values in some controls on at least some Qualcomm devices (this was a wcd9340) which clearly indicates that there's some problems that need fixing. This isn't helped by the fact that snd_soc_info_volsw_sx() is implemented by calling snd_soc_info_volsw() and then applying an offset to the maximum value which makes things harder to follow - I think we should make snd_soc_info_volsw_sx() a standalone function, and we probably need some fixes in some combination of the shared functions and the drivers as well. Can people please check how the values in the current drivers correspond to what the actual bitfields in the registers have so we can get a handle on what's going on here? I've pasted a quick grep for _SX controls below: sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC FIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC IIR Output Shift", sound/soc/codecs/ab8500-codec.c: SOC_SINGLE_XR_SX("ANC Warp Delay", sound/soc/codecs/cs35l33.c: SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, sound/soc/codecs/cs35l34.c: SOC_SINGLE_SX_TLV("Digital Volume", CS35L34_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Audio Volume", CS35L35_AMP_DIG_VOL, sound/soc/codecs/cs35l35.c: SOC_SINGLE_SX_TLV("Digital Advisory Volume", CS35L35_ADV_DIG_VOL, sound/soc/codecs/cs35l36.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L36_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs35l41.c: SOC_SINGLE_SX_TLV("Digital PCM Volume", CS35L41_AMP_DIG_VOL_CTRL, sound/soc/codecs/cs4265.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL, sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", sound/soc/codecs/cs42l51.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L52_MASTERA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, sound/soc/codecs/cs42l52.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", sound/soc/codecs/cs42l52.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, sound/soc/codecs/cs42l56.c: SOC_SINGLE_SX_TLV("Beep Volume", CS42L56_BEEP_FREQ_OFFTIME, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, sound/soc/codecs/cs42l73.c: SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Speakerphone Digital Volume", sound/soc/codecs/cs42l73.c: SOC_SINGLE_SX_TLV("Ear Speaker Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B PGA Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC1B Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2A Digital Volume", sound/soc/codecs/cs53l30.c: SOC_SINGLE_SX_TLV("ADC2B Digital Volume", sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP1 Volume", LPASS_CDC_IIR1_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP2 Volume", LPASS_CDC_IIR1_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP3 Volume", LPASS_CDC_IIR1_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR1 INP4 Volume", LPASS_CDC_IIR1_GAIN_B4_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP1 Volume", LPASS_CDC_IIR2_GAIN_B1_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP2 Volume", LPASS_CDC_IIR2_GAIN_B2_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP3 Volume", LPASS_CDC_IIR2_GAIN_B3_CTL, sound/soc/codecs/msm8916-wcd-digital.c: SOC_SINGLE_SX_TLV("IIR2 INP4 Volume", LPASS_CDC_IIR2_GAIN_B4_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Digital Volume", WCD9335_CDC_RX0_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Digital Volume", WCD9335_CDC_RX1_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Digital Volume", WCD9335_CDC_RX2_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Digital Volume", WCD9335_CDC_RX3_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Digital Volume", WCD9335_CDC_RX4_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Digital Volume", WCD9335_CDC_RX5_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Digital Volume", WCD9335_CDC_RX6_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Digital Volume", WCD9335_CDC_RX7_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Digital Volume", WCD9335_CDC_RX8_RX_VOL_CTL, sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX0 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX1 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX2 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX3 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX4 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX5 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX6 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX7 Mix Digital Volume", sound/soc/codecs/wcd9335.c: SOC_SINGLE_SX_TLV("RX8 Mix Digital Volume", Thanks, Mark

3 4

"topology: nhlt plugin: second attempt to fix the .so linking" cause build static library failed
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-utils issue #152 was opened from youling257: ./configure --prefix=/system/usr --with-alsa-inc-prefix=/system/usr/include --with-alsa-prefix=/system/usr/lib --enable-static=yes --enable-shared=no make LDFLAGS="--static" -j8 Revert "topology: nhlt plugin: second attempt to fix the .so linking" can build static library. Issue URL : https://github.com/alsa-project/alsa-utils/issues/152 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: --- Add remove to put PM usage counter. v3: --- Modify commit message. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

1 0

[PATCH V2] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 02 Jun '22

02 Jun '22

a) Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. b) Add remove for missing pm_runtime_disable(). v2: add remove to put PM usage counter. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..55e640cba87d 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } @@ -1242,6 +1244,12 @@ static int fsl_xcvr_probe(struct platform_device *pdev) return ret; } +static int fsl_xcvr_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + return 0; +} + static __maybe_unused int fsl_xcvr_runtime_suspend(struct device *dev) { struct fsl_xcvr *xcvr = dev_get_drvdata(dev); @@ -1370,6 +1378,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = &fsl_xcvr_pm_ops, .of_match_table = fsl_xcvr_dt_ids, }, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); -- 2.31.1

2 1

1.2.7: missing git tag
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-utils issue #151 was opened from kloczek: https://www.alsa-project.org/wiki/Main_Page says that last release is 1.2.7 but there is no git tag for that version here. Issue URL : https://github.com/alsa-project/alsa-utils/issues/151 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

1.2.7: missing git tag
by GitHub issues - opened 02 Jun '22

02 Jun '22

alsa-project/alsa-lib issue #232 was opened from kloczek: https://www.alsa-project.org/wiki/Main_Page says that last release is 1.2.7 but there is no git tag for that version here. Issue URL : https://github.com/alsa-project/alsa-lib/issues/232 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

soundwire: device drivers for PC based hardware
by Muni Sekhar 01 Jun '22

01 Jun '22

Hi all, I am looking for the soundwire device drivers which run on x86 CPU for PC based platforms. linux/drivers/soundwire at master · torvalds/linux · GitHub Can the above mentioned mainline tree drivers can be used for this purpose? config SOUNDWIRE_INTEL depends on ACPI && SND_SOC config SOUNDWIRE_QCOM depends on SND_SOC As per the Kconfig file, both the Intel and Qualcomm SoundWire Master driver depend on SND_SOC, does it mean it is only possible to run on SoC CPU(Arm chipsets)? -- Thanks, Sekhar

2 1

Re: [syzbot] KASAN: use-after-free Read in snd_rawmidi_transmit
by syzbot 01 Jun '22

01 Jun '22

syzbot has bisected this issue to: commit d098538ed4e8a6c09f86cf243f406c1451066040 Author: Sasha Neftin <sasha.neftin(a)intel.com> Date: Mon Mar 21 06:05:10 2022 +0000 igc: Remove igc_set_spd_dplx method bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=16458313f00000 start commit: 2a5699b0de4e Merge tag 'leds-5.19-rc1' of git://git.kernel.. git tree: upstream final oops: https://syzkaller.appspot.com/x/report.txt?x=15458313f00000 console output: https://syzkaller.appspot.com/x/log.txt?x=11458313f00000 kernel config: https://syzkaller.appspot.com/x/.config?x=f915857c44e13980 dashboard link: https://syzkaller.appspot.com/bug?extid=39e3268af9968f153591 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=119d1fcbf00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17716bedf00000 Reported-by: syzbot+39e3268af9968f153591(a)syzkaller.appspotmail.com Fixes: d098538ed4e8 ("igc: Remove igc_set_spd_dplx method") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

1 0

[PATCH AUTOSEL 5.15 14/37] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..5461cdf907e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1824,6 +1826,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 13/37] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 65f61695f561..5d093fb4896b 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2237,7 +2237,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2279,6 +2279,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * 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 */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 17/48] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..5461cdf907e2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1824,6 +1826,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 16/48] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 34888d4fc4a8..21ab5e6f7062 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2246,7 +2246,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2288,6 +2288,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * 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 */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 06/48] ASoC: amd: Add driver data to acp6x machine driver
by Sasha Levin 01 Jun '22

01 Jun '22

From: Mario Limonciello <mario.limonciello(a)amd.com> [ Upstream commit e521f087780d07731e8c950f2f34d08358c86bc9 ] Currently all of the quirked systems use the same card and so the DMI quirk list doesn't contain driver data. Add driver data to these quirks and then check the data was present or not. This will allow potentially setting quirks for systems with faulty firmware that claims to have a DMIC but doesn't really. Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://lore.kernel.org/r/20220411134532.13538-2-mario.limonciello@amd.com 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 | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 9a767f47b89f..959b70e8baf2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,108 +45,126 @@ 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, "21D2"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D3"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D4"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D5"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CF"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CG"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CR"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AW"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AX"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BN"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CH"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CJ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CK"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D8"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D9"), @@ -157,18 +175,21 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { static int acp6x_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; int ret; - const struct dmi_system_id *dmi_id; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); - if (!dmi_id) + if (dmi_id) + platform_set_drvdata(pdev, dmi_id->driver_data); + + card = platform_get_drvdata(pdev); + if (!card) return -ENODEV; - card = &acp6x_card; acp6x_card.dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 17/49] ALSA: usb-audio: Move generic implicit fb quirk entries into quirks.c
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 67d64069bc0867e52e73a1e255b17462005ca9b4 ] Use the new quirk bits to manage the generic implicit fb quirk entries. This makes easier to compare with other devices. Link: https://lore.kernel.org/r/20220421064101.12456-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 5 ----- sound/usb/quirks.c | 6 ++++++ 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 1fd087128538..e1bf1b5da423 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -45,11 +45,6 @@ struct snd_usb_implicit_fb_match { /* Implicit feedback quirk table for playback */ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { - /* Generic matching */ - IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ - /* Fixed EP */ /* FIXME: check the availability of generic matching */ IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fbbe59054c3f..e8468f9b007d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1793,6 +1793,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x046d, 0x09a4, /* Logitech QuickCam E 3500 */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0499, 0x1509, /* Steinberg UR22 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ @@ -1826,6 +1828,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0763, 0x2030, /* M-Audio Fast Track C400 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0763, 0x2031, /* M-Audio Fast Track C600 */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x0951, 0x16ad, /* Kingston HyperX */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 16/49] ALSA: usb-audio: Add quirk bits for enabling/disabling generic implicit fb
by Sasha Levin 01 Jun '22

01 Jun '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 0f1f7a6661394fe4a53db254c346d6aa2dd64397 ] For making easier to test, add the new quirk_flags bits 17 and 18 to enable and disable the generic implicit feedback mode. The bit 17 is equivalent with implicit_fb=1 option, applying the generic implicit feedback sync mode. OTOH, the bit 18 disables the implicit fb mode forcibly. Link: https://lore.kernel.org/r/20220421064101.12456-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- Documentation/sound/alsa-configuration.rst | 4 +++- sound/usb/implicit.c | 5 ++++- sound/usb/usbaudio.h | 6 ++++++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 34888d4fc4a8..21ab5e6f7062 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2246,7 +2246,7 @@ implicit_fb Apply the generic implicit feedback sync mode. When this is set and the playback stream sync mode is ASYNC, the driver tries to tie an adjacent ASYNC capture stream as the implicit feedback - source. + source. This is equivalent with quirk_flags bit 17. use_vmalloc Use vmalloc() for allocations of the PCM buffers (default: yes). For architectures with non-coherent memory like ARM or MIPS, the @@ -2288,6 +2288,8 @@ quirk_flags * bit 14: Ignore errors for mixer access * bit 15: Support generic DSD raw U32_BE format * bit 16: Set up the interface at first like UAC1 + * bit 17: Apply the generic implicit feedback sync mode + * bit 18: Don't apply implicit feedback sync mode This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 2d444ec74202..1fd087128538 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -350,7 +350,8 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Try the generic implicit fb if available */ - if (chip->generic_implicit_fb) + if (chip->generic_implicit_fb || + (chip->quirk_flags & QUIRK_FLAG_GENERIC_IMPLICIT_FB)) return add_generic_implicit_fb(chip, fmt, alts); /* No quirk */ @@ -387,6 +388,8 @@ int snd_usb_parse_implicit_fb_quirk(struct snd_usb_audio *chip, struct audioformat *fmt, struct usb_host_interface *alts) { + if (chip->quirk_flags & QUIRK_FLAG_SKIP_IMPLICIT_FB) + return 0; if (fmt->endpoint & USB_DIR_IN) return audioformat_capture_quirk(chip, fmt, alts); else diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b8359a0aa008..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -164,6 +164,10 @@ extern bool snd_usb_skip_validation; * Support generic DSD raw U32_BE format * QUIRK_FLAG_SET_IFACE_FIRST: * Set up the interface at first like UAC1 + * QUIRK_FLAG_GENERIC_IMPLICIT_FB + * 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 */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -183,5 +187,7 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_IGNORE_CTL_ERROR (1U << 14) #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +#define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) +#define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 06/49] ASoC: amd: Add driver data to acp6x machine driver
by Sasha Levin 01 Jun '22

01 Jun '22

From: Mario Limonciello <mario.limonciello(a)amd.com> [ Upstream commit e521f087780d07731e8c950f2f34d08358c86bc9 ] Currently all of the quirked systems use the same card and so the DMI quirk list doesn't contain driver data. Add driver data to these quirks and then check the data was present or not. This will allow potentially setting quirks for systems with faulty firmware that claims to have a DMIC but doesn't really. Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> Link: https://lore.kernel.org/r/20220411134532.13538-2-mario.limonciello@amd.com 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 | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 9a767f47b89f..959b70e8baf2 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -45,108 +45,126 @@ 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, "21D2"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D3"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D4"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D5"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CF"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CG"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CR"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AW"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21AX"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BN"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21BQ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CH"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CJ"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CK"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D8"), } }, { + .driver_data = &acp6x_card, .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), DMI_MATCH(DMI_PRODUCT_NAME, "21D9"), @@ -157,18 +175,21 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { static int acp6x_probe(struct platform_device *pdev) { + const struct dmi_system_id *dmi_id; struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; int ret; - const struct dmi_system_id *dmi_id; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); - if (!dmi_id) + if (dmi_id) + platform_set_drvdata(pdev, dmi_id->driver_data); + + card = platform_get_drvdata(pdev); + if (!card) return -ENODEV; - card = &acp6x_card; acp6x_card.dev = &pdev->dev; - platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { -- 2.35.1

1 0

[PATCH] ASoC: fsl: Check before clk_put() not needed
by Yihao Han 01 Jun '22

01 Jun '22

clk_put() already checks the clk ptr using !clk and IS_ERR() so there is no need to check it again before calling it. Signed-off-by: Yihao Han <hanyihao(a)vivo.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 580a0d963f0e..16a281820186 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -185,8 +185,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) put_device: put_device(&codec_dev->dev); fail: - if (data && !IS_ERR(data->codec_clk)) - clk_put(data->codec_clk); + clk_put(data->codec_clk); of_node_put(ssi_np); of_node_put(codec_np); -- 2.17.1

2 1

[PATCH] ASoC: Intel: cirrus-common: fix incorrect channel mapping
by Brent Lu 01 Jun '22

01 Jun '22

From: xliu <xiang.liu(a)cirrus.com> The default mapping of ASPRX1 Slot is left channel. Map the slots of right amplifiers (WR and TR) to right channel. Signed-off-by: xliu <xiang.liu(a)cirrus.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index e71d74ec1b0b..64ca0e3991dc 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -107,6 +107,7 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; int clk_freq, i, ret; + int rx_ch[2] = {1, 0}; clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ @@ -134,6 +135,17 @@ static int cs35l41_hw_params(struct snd_pcm_substream *substream, ret); return ret; } + + /* Setup for R channel Slot: WR and TR */ + if (i % 2) { + ret = snd_soc_dai_set_channel_map(codec_dai, 0, NULL, + ARRAY_SIZE(rx_ch), rx_ch); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set channel map, ret %d\n", + ret); + return ret; + } + } } return 0; -- 2.25.1

3 2

[PATCH] ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback
by Srinivasa Rao Mandadapu 01 Jun '22

01 Jun '22

Replace page protection permissions from noncashed to writecombine, in lpass codec DMA path mmp callabck, to support 64 bit chromeOS. Avoid SIGBUS error in userspace caused by noncached permissions in 64 bit chromeOS. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index f03a7ae..b41ab7a 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -898,7 +898,7 @@ static int lpass_platform_cdc_dma_mmap(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; unsigned long size, offset; - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); size = vma->vm_end - vma->vm_start; offset = vma->vm_pgoff << PAGE_SHIFT; return io_remap_pfn_range(vma, vma->vm_start, -- 2.7.4

2 1

[PATCH 0/2] Add pinctrl support adsp based platforms
by Srinivasa Rao Mandadapu 01 Jun '22

01 Jun '22

This patch set is to add pinctrl support adsp enabled sc7280 platforms. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add compatible string for adsp based platforms pinctrl: qcom: sc7280: Add lpi pinctrl variant data structure for adsp based targets .../bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 4 +++- drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 14 ++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) -- 2.7.4

3 5

Re: [syzbot] KASAN: use-after-free Read in snd_rawmidi_transmit
by syzbot 01 Jun '22

01 Jun '22

syzbot has found a reproducer for the following issue on: HEAD commit: 2a5699b0de4e Merge tag 'leds-5.19-rc1' of git://git.kernel.. git tree: upstream console+strace: https://syzkaller.appspot.com/x/log.txt?x=13df4d13f00000 kernel config: https://syzkaller.appspot.com/x/.config?x=f915857c44e13980 dashboard link: https://syzkaller.appspot.com/bug?extid=39e3268af9968f153591 compiler: Debian clang version 13.0.1-++20220126092033+75e33f71c2da-1~exp1~20220126212112.63, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=119d1fcbf00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17716bedf00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+39e3268af9968f153591(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1286 [inline] BUG: KASAN: use-after-free in snd_rawmidi_transmit+0x3f3/0x420 sound/core/rawmidi.c:1411 Read of size 1 at addr ffff888074d3c07f by task kworker/1:1H/756 CPU: 1 PID: 756 Comm: kworker/1:1H Not tainted 5.18.0-syzkaller-11650-g2a5699b0de4e #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_usbmidi_out_work Call Trace: <TASK> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0x1e3/0x2cb lib/dump_stack.c:106 print_address_description+0x65/0x4b0 mm/kasan/report.c:313 print_report+0xf4/0x210 mm/kasan/report.c:429 kasan_report+0xfb/0x130 mm/kasan/report.c:491 __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1286 [inline] snd_rawmidi_transmit+0x3f3/0x420 sound/core/rawmidi.c:1411 snd_usbmidi_standard_output+0x202/0xf30 sound/usb/midi.c:650 snd_usbmidi_do_output+0x20e/0x530 sound/usb/midi.c:311 process_one_work+0x81c/0xd10 kernel/workqueue.c:2289 worker_thread+0xb14/0x1330 kernel/workqueue.c:2436 kthread+0x266/0x300 kernel/kthread.c:376 ret_from_fork+0x1f/0x30 </TASK> Allocated by task 3983: kasan_save_stack mm/kasan/common.c:38 [inline] kasan_set_track mm/kasan/common.c:45 [inline] set_alloc_info mm/kasan/common.c:436 [inline] ____kasan_kmalloc+0xdc/0x110 mm/kasan/common.c:515 kasan_kmalloc include/linux/kasan.h:234 [inline] __kmalloc_node+0x262/0x400 mm/slub.c:4465 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x6e/0x160 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] kvzalloc include/linux/slab.h:758 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:162 [inline] open_substream+0x2a1/0x6d0 sound/core/rawmidi.c:306 rawmidi_open_priv+0xc2/0x6a0 sound/core/rawmidi.c:357 snd_rawmidi_kernel_open+0x1e3/0x280 sound/core/rawmidi.c:392 midisynth_use+0xf9/0x2b0 sound/core/seq/seq_midi.c:215 subscribe_port sound/core/seq/seq_ports.c:412 [inline] check_and_subscribe_port+0x633/0xac0 sound/core/seq/seq_ports.c:495 snd_seq_port_connect+0x27e/0x480 sound/core/seq/seq_ports.c:581 snd_seq_ioctl_subscribe_port+0x394/0x750 sound/core/seq/seq_clientmgr.c:1492 snd_seq_oss_midi_open+0x44b/0x9b0 sound/core/seq/oss/seq_oss_midi.c:359 snd_seq_oss_synth_setup_midi+0x126/0x530 sound/core/seq/oss/seq_oss_synth.c:269 snd_seq_oss_open+0x96d/0x1020 sound/core/seq/oss/seq_oss_init.c:260 odev_open+0x5e/0x90 sound/core/seq/oss/seq_oss.c:128 chrdev_open+0x5fb/0x680 fs/char_dev.c:414 do_dentry_open+0x789/0x1040 fs/open.c:848 do_open fs/namei.c:3527 [inline] path_openat+0x26c0/0x2ec0 fs/namei.c:3660 do_filp_open+0x277/0x4f0 fs/namei.c:3687 do_sys_openat2+0x13b/0x500 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x243/0x290 fs/open.c:1305 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 Freed by task 3983: kasan_save_stack mm/kasan/common.c:38 [inline] kasan_set_track+0x4c/0x70 mm/kasan/common.c:45 kasan_set_free_info+0x1f/0x40 mm/kasan/generic.c:370 ____kasan_slab_free+0xd8/0x110 mm/kasan/common.c:366 kasan_slab_free include/linux/kasan.h:200 [inline] slab_free_hook mm/slub.c:1727 [inline] slab_free_freelist_hook+0x12e/0x1a0 mm/slub.c:1753 slab_free mm/slub.c:3507 [inline] kfree+0xc6/0x210 mm/slub.c:4555 snd_rawmidi_runtime_free sound/core/rawmidi.c:176 [inline] close_substream+0x38a/0x5f0 sound/core/rawmidi.c:528 rawmidi_release_priv+0xf1/0x180 sound/core/rawmidi.c:547 snd_rawmidi_kernel_release+0x42/0xa0 sound/core/rawmidi.c:564 unsubscribe_port sound/core/seq/seq_ports.c:437 [inline] __delete_and_unsubscribe_port+0x15c/0x330 sound/core/seq/seq_ports.c:537 snd_seq_port_disconnect+0x533/0x660 sound/core/seq/seq_ports.c:616 snd_seq_ioctl_unsubscribe_port+0x394/0x750 sound/core/seq/seq_clientmgr.c:1537 snd_seq_oss_midi_close+0x33f/0x740 sound/core/seq/oss/seq_oss_midi.c:404 snd_seq_oss_synth_reset+0x3b6/0x9d0 sound/core/seq/oss/seq_oss_synth.c:406 snd_seq_oss_reset+0x5b/0x240 sound/core/seq/oss/seq_oss_init.c:435 snd_seq_oss_release+0xe9/0x300 sound/core/seq/oss/seq_oss_init.c:412 odev_release+0x52/0x70 sound/core/seq/oss/seq_oss.c:144 __fput+0x3b9/0x820 fs/file_table.c:317 task_work_run+0x146/0x1c0 kernel/task_work.c:177 ptrace_notify+0x29f/0x350 kernel/signal.c:2372 ptrace_report_syscall include/linux/ptrace.h:427 [inline] ptrace_report_syscall_exit include/linux/ptrace.h:489 [inline] syscall_exit_work+0x8c/0xe0 kernel/entry/common.c:249 syscall_exit_to_user_mode_prepare+0x62/0x80 kernel/entry/common.c:276 __syscall_exit_to_user_mode_work kernel/entry/common.c:281 [inline] syscall_exit_to_user_mode+0xa/0x60 kernel/entry/common.c:294 entry_SYSCALL_64_after_hwframe+0x46/0xb0 The buggy address belongs to the object at ffff888074d3c000 which belongs to the cache kmalloc-4k of size 4096 The buggy address is located 127 bytes inside of 4096-byte region [ffff888074d3c000, ffff888074d3d000) The buggy address belongs to the physical page: page:ffffea0001d34e00 refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x74d38 head:ffffea0001d34e00 order:3 compound_mapcount:0 compound_pincount:0 flags: 0xfff00000010200(slab|head|node=0|zone=1|lastcpupid=0x7ff) raw: 00fff00000010200 0000000000000000 dead000000000122 ffff888011442140 raw: 0000000000000000 0000000000040004 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 3, migratetype Unmovable, gfp_mask 0xd2040(__GFP_IO|__GFP_NOWARN|__GFP_NORETRY|__GFP_COMP|__GFP_NOMEMALLOC), pid 3983, tgid 3982 (syz-executor242), ts 217737880306, free_ts 217582307012 prep_new_page mm/page_alloc.c:2456 [inline] get_page_from_freelist+0x72b/0x7a0 mm/page_alloc.c:4198 __alloc_pages+0x259/0x560 mm/page_alloc.c:5426 alloc_slab_page+0x70/0xf0 mm/slub.c:1797 allocate_slab+0x5e/0x520 mm/slub.c:1942 new_slab mm/slub.c:2002 [inline] ___slab_alloc+0x41e/0xcd0 mm/slub.c:3002 __slab_alloc mm/slub.c:3089 [inline] slab_alloc_node mm/slub.c:3180 [inline] slab_alloc mm/slub.c:3222 [inline] __kmalloc+0x2ba/0x370 mm/slub.c:4413 kmalloc include/linux/slab.h:605 [inline] tomoyo_realpath_from_path+0xd8/0x5f0 security/tomoyo/realpath.c:254 tomoyo_get_realpath security/tomoyo/file.c:151 [inline] tomoyo_check_open_permission+0x22f/0x490 security/tomoyo/file.c:771 security_file_open+0x50/0x560 security/security.c:1645 do_dentry_open+0x34a/0x1040 fs/open.c:835 do_open fs/namei.c:3527 [inline] path_openat+0x26c0/0x2ec0 fs/namei.c:3660 do_filp_open+0x277/0x4f0 fs/namei.c:3687 do_sys_openat2+0x13b/0x500 fs/open.c:1278 do_sys_open fs/open.c:1294 [inline] __do_sys_openat fs/open.c:1310 [inline] __se_sys_openat fs/open.c:1305 [inline] __x64_sys_openat+0x243/0x290 fs/open.c:1305 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 page last free stack trace: reset_page_owner include/linux/page_owner.h:24 [inline] free_pages_prepare mm/page_alloc.c:1371 [inline] free_pcp_prepare+0x812/0x900 mm/page_alloc.c:1421 free_unref_page_prepare mm/page_alloc.c:3343 [inline] free_unref_page+0x7d/0x390 mm/page_alloc.c:3438 free_slab mm/slub.c:2041 [inline] discard_slab mm/slub.c:2047 [inline] __unfreeze_partials+0x1ab/0x200 mm/slub.c:2521 put_cpu_partial+0x116/0x180 mm/slub.c:2597 do_slab_free mm/slub.c:3495 [inline] ___cache_free+0x118/0x1a0 mm/slub.c:3514 qlist_free_all+0x2b/0x70 mm/kasan/quarantine.c:187 kasan_quarantine_reduce+0x169/0x180 mm/kasan/quarantine.c:294 __kasan_slab_alloc+0x2f/0xe0 mm/kasan/common.c:446 kasan_slab_alloc include/linux/kasan.h:224 [inline] slab_post_alloc_hook mm/slab.h:750 [inline] slab_alloc_node mm/slub.c:3214 [inline] __kmalloc_node+0x1f3/0x400 mm/slub.c:4461 kmalloc_node include/linux/slab.h:623 [inline] kvmalloc_node+0x6e/0x160 mm/util.c:613 kvmalloc include/linux/slab.h:750 [inline] seq_buf_alloc fs/seq_file.c:38 [inline] seq_read_iter+0x1f6/0xd30 fs/seq_file.c:210 call_read_iter include/linux/fs.h:2054 [inline] new_sync_read fs/read_write.c:401 [inline] vfs_read+0xa01/0xd10 fs/read_write.c:482 ksys_read+0x19b/0x2c0 fs/read_write.c:620 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x2b/0x70 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x46/0xb0 Memory state around the buggy address: ffff888074d3bf00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff888074d3bf80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff888074d3c000: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff888074d3c080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff888074d3c100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ==================================================================

1 0

[GIT PULL] ASoC fixes for v5.19-rc0
by Mark Brown 01 Jun '22

01 Jun '22

The following changes since commit e5cd20e0d6713138444cc3f3f982712cf9a36143: ASoC: SOF: Introduce generic (in)firmware tracing infrastructure (2022-05-20 14:56:35 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc0 for you to fetch changes up to ab0925ab728163b854ea17da09e299254a86076e: ASoC: da7219: Small fixes for jack detection and removal (2022-05-31 10:16:59 +0200) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more fixes that came in during the merge window - nothing huge here, there is one core fix for DPCM from Pierre but mostly driver changes. ---------------------------------------------------------------- Adam Thomson (2): ASoC: da7219: Fix pole orientation detection on certain headsets ASoC: da7219: cancel AAD related work earlier for jack removal David Lin (1): ASoC: Intel: common: fix typo for tplg naming Mark Brown (1): ASoC: da7219: Small fixes for jack detection and removal Oder Chiou (1): ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM Pierre-Louis Bossart (1): ASoC: soc-pcm: fix BE transition for TRIGGER_START Shengjiu Wang (1): ASoC: fsl_sai: Fix FSL_SAI_xDR/xFR definition Syed Saba kareem (1): ASoC: SOF: amd: Fixed Build error Uwe Kleine-König (1): ASoC: Intel: avs: Fix build error on arc, m68k and sparc sound/soc/amd/acp/acp-pci.c | 1 + sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++--- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/fsl/fsl_sai.h | 4 ++-- sound/soc/intel/avs/board_selection.c | 3 ++- sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- sound/soc/soc-pcm.c | 2 -- 9 files changed, 34 insertions(+), 11 deletions(-)

2 1

[PATCH 0/5] seq: add interface for queue timer
by GitHub pull_request - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-gobject pull request #72 was opened from takaswie: Current implementation uses function of QueueTimer to retrieve data for variants of queue timer, however it's not enough convenient in the case of language bindings. The usage of GObject.Value in object property is expected to be supported widely instead. This patchset uses object classes derived from GObject to express the variants. The common properties are expressed by QueueTimerCommon interface. The ALSATimer variant is currently supported to implement the interface. ``` Takashi Sakamoto (5): doc: seq: fix URL to document for ALSATimer seq: queue-timer-common: add common interface for queue timer seq: queue-timer-alsa: add class for queue timer specialized to ALSA Timer seq: user-client: use QueueTimerCommon interface instead of QueueTimer seq: queue-timer/queue-timer-data-alsa: remove unused gobject classes doc/seq/alsaseq.toml.in | 2 +- doc/seq/urlmap.js | 2 +- src/seq/alsaseq.h | 5 +- src/seq/alsaseq.map | 16 +- src/seq/meson.build | 8 +- src/seq/privates.h | 21 ++- src/seq/queue-timer-alsa.c | 134 +++++++++++++++++ src/seq/queue-timer-alsa.h | 22 +++ src/seq/queue-timer-common.c | 87 +++++++++++ src/seq/queue-timer-common.h | 20 +++ src/seq/queue-timer-data-alsa.c | 76 ---------- src/seq/queue-timer-data-alsa.h | 32 ---- src/seq/queue-timer.c | 138 ------------------ src/seq/queue-timer.h | 26 ---- src/seq/user-client.c | 42 +++--- src/seq/user-client.h | 4 +- ...q-queue-timer => alsaseq-queue-timer-alsa} | 8 +- tests/meson.build | 2 +- 18 files changed, 325 insertions(+), 320 deletions(-) create mode 100644 src/seq/queue-timer-alsa.c create mode 100644 src/seq/queue-timer-alsa.h create mode 100644 src/seq/queue-timer-common.c create mode 100644 src/seq/queue-timer-common.h delete mode 100644 src/seq/queue-timer-data-alsa.c delete mode 100644 src/seq/queue-timer-data-alsa.h delete mode 100644 src/seq/queue-timer.c delete mode 100644 src/seq/queue-timer.h rename tests/{alsaseq-queue-timer => alsaseq-queue-timer-alsa} (76%) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/72 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/72.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 00/13] ctl: add interfaces for element information
by GitHub pull_request - opened 01 Jun '22

01 Jun '22

alsa-project/alsa-gobject pull request #71 was opened from takaswie: Current implementation uses function of ElemInfo to express variants of element information, however it's not enough convenient in the case of language bindings. The usage of GObject.Value in object property is expected to be supported widely instead. This patchset uses object classes derived from GObject to express the variants. The common properties are expressed by two interfaces; ElemInfoCommon and ElemInfoSingleArray. All of classes implement the former, and the latter is implemented except for IEC60958 element information. ``` Takashi Sakamoto (13): ctl: card: fix indentation fix missing usage of declaration macro ctl: card: add error path when generating name list at element addition ctl: elem-info-common: add common interface for element information ctl: elem-info-iec60958: add class for element information of IEC60958 type ctl: elem-info-single-array: add interface for single array element information ctl: elem-info-boolean: add class for element information of boolean type ctl: elem-info-bytes: add class for element information of bytes type ctl: elem-info-integer: add class for element information of integer type ctl: elem-info-integer64: add class for element information of integer64 type ctl: elem-info-enumerated: add class for element information of enumerated type ctl: card: use ALSACtl.ElemInfoCommon instead of ElemInfo ctl: elem-info: remove unused classes samples/ctl | 56 +-- src/ctl/alsactl-enum-types.h | 4 + src/ctl/alsactl.h | 10 +- src/ctl/alsactl.map | 30 +- src/ctl/card.c | 156 ++++--- src/ctl/card.h | 6 +- src/ctl/elem-info-boolean.c | 90 +++++ src/ctl/elem-info-boolean.h | 22 + src/ctl/elem-info-bytes.c | 89 ++++ src/ctl/elem-info-bytes.h | 21 + src/ctl/elem-info-common.c | 135 +++++++ src/ctl/elem-info-common.h | 20 + src/ctl/elem-info-enumerated.c | 137 +++++++ src/ctl/elem-info-enumerated.h | 22 + src/ctl/elem-info-iec60958.c | 80 ++++ src/ctl/elem-info-iec60958.h | 22 + src/ctl/elem-info-integer.c | 162 ++++++++ src/ctl/elem-info-integer.h | 22 + src/ctl/elem-info-integer64.c | 162 ++++++++ src/ctl/elem-info-integer64.h | 22 + src/ctl/elem-info-single-array.c | 68 ++++ src/ctl/elem-info-single-array.h | 20 + src/ctl/elem-info.c | 379 ------------------ src/ctl/elem-info.h | 30 -- src/ctl/meson.build | 18 +- src/ctl/privates.h | 54 ++- src/hwdep/alsahwdep-enum-types.h | 4 + src/rawmidi/alsarawmidi-enum-types.h | 4 + src/seq/alsaseq-enum-types.h | 4 + src/timer/alsatimer-enum-types.h | 4 + ...tl-elem-info => alsactl-elem-info-boolean} | 10 +- tests/alsactl-elem-info-bytes | 26 ++ tests/alsactl-elem-info-enumerated | 27 ++ tests/alsactl-elem-info-iec60958 | 25 ++ tests/alsactl-elem-info-integer | 29 ++ tests/alsactl-elem-info-integer64 | 29 ++ tests/meson.build | 7 +- 37 files changed, 1492 insertions(+), 514 deletions(-) create mode 100644 src/ctl/elem-info-boolean.c create mode 100644 src/ctl/elem-info-boolean.h create mode 100644 src/ctl/elem-info-bytes.c create mode 100644 src/ctl/elem-info-bytes.h create mode 100644 src/ctl/elem-info-common.c create mode 100644 src/ctl/elem-info-common.h create mode 100644 src/ctl/elem-info-enumerated.c create mode 100644 src/ctl/elem-info-enumerated.h create mode 100644 src/ctl/elem-info-iec60958.c create mode 100644 src/ctl/elem-info-iec60958.h create mode 100644 src/ctl/elem-info-integer.c create mode 100644 src/ctl/elem-info-integer.h create mode 100644 src/ctl/elem-info-integer64.c create mode 100644 src/ctl/elem-info-integer64.h create mode 100644 src/ctl/elem-info-single-array.c create mode 100644 src/ctl/elem-info-single-array.h delete mode 100644 src/ctl/elem-info.c delete mode 100644 src/ctl/elem-info.h rename tests/{alsactl-elem-info => alsactl-elem-info-boolean} (68%) create mode 100644 tests/alsactl-elem-info-bytes create mode 100644 tests/alsactl-elem-info-enumerated create mode 100644 tests/alsactl-elem-info-iec60958 create mode 100644 tests/alsactl-elem-info-integer create mode 100644 tests/alsactl-elem-info-integer64 ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/71 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/71.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

ucm2: rk3399-gru-sound: Add missing symlink from conf.d tree
by GitHub pull_request - opened 31 May '22

31 May '22

alsa-project/alsa-ucm-conf pull request #162 was opened from alpernebbi: Since V2Module lookup was disabled my rk3399-gru-kevin chromebook can't find the UCM it's supposed to use. This adds a `conf.d/rk3399-gru-soun/rk3399-gru-soun.conf` symlink for it. The name is not a typo: $ cat /proc/asound/cards 0 [rk3399grusound ]: rk3399-gru-soun - rk3399-gru-sound Unknown-UnknownProduct- $ strace -e trace=file pulseaudio 2>&1 | grep -i ucm2 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", R_OK) = 0 openat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", O_RDONLY) = 16 openat(AT_FDCWD, "/usr/share/alsa/ucm2/lib/generic.conf", O_RDONLY) = 16 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/Unknown-UnknownProduct-.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/rk3399-gru-soun.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", R_OK) = 0 openat(AT_FDCWD, "/usr/share/alsa/ucm2/ucm.conf", O_RDONLY) = 16 openat(AT_FDCWD, "/usr/share/alsa/ucm2/lib/generic.conf", O_RDONLY) = 16 faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/Unknown-UnknownProduct-.conf", R_OK) = -1 ENOENT (No such file or directory) faccessat(AT_FDCWD, "/usr/share/alsa/ucm2/conf.d/rk3399-gru-soun/rk3399-gru-soun.conf", R_OK) = -1 ENOENT (No such file or directory) (Ignore the `Unknown-UnknownProduct-`, that's because I'm using a custom unfinished U-Boot build. For more info there's an [older alsa-info output](https://alsa-project.org/db/?f=9b4102975fa9c0f109af6ad687c856c620d7… I had uploaded) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/162 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/162.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALSA 1.2.7 release
by Jaroslav Kysela 31 May '22

31 May '22

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf pyalsa Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.6.3_v1.2.7 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH] selftests: alsa: Handle pkg-config failure more gracefully
by Mark Brown 31 May '22

31 May '22

Follow the pattern used by other selftests like memfd and fall back on the standard toolchain options to build with a system installed alsa-lib if we don't get anything from pkg-config. This reduces our build dependencies a bit in the common case while still allowing use of pkg-config in case there is a need for it. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- tools/testing/selftests/alsa/Makefile | 3 +++ 1 file changed, 3 insertions(+) diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index f64d9090426d..fd8ddce2b1a6 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -3,6 +3,9 @@ CFLAGS += $(shell pkg-config --cflags alsa) LDLIBS += $(shell pkg-config --libs alsa) +ifeq ($(LDLIBS),) +LDLIBS += -lasound +endif TEST_GEN_PROGS := mixer-test -- 2.30.2

2 1

[PATCH] ALSA: usb-audio: Optimize TEAC clock quirk
by Takashi Iwai 31 May '22

31 May '22

Maris found out that the quirk for TEAC devices to work around the clock setup is needed to apply only when the base clock is changed, e.g. from 48000-based clocks (48000, 96000, 192000, 384000) to 44100-based clocks (44100, 88200, 176400, 352800), or vice versa, while switching to another clock with the same base clock doesn't need the (forcible) interface setup. This patch implements the optimization for the TEAC clock quirk to avoid the unnecessary interface re-setup. Fixes: 5ce0b06ae5e6 ("ALSA: usb-audio: Workaround for clock setup on TEAC devices") Reported-by: Maris Abele <maris7abele(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 3c435d379306..db429320599e 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -573,10 +573,14 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, } /* FIXME - TEAC devices require the immediate interface setup */ - if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { - usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); - if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) - msleep(50); + if (USB_ID_VENDOR(chip->usb_id) == 0x0644) { + bool cur_base_48k = (rate % 48000 == 0); + bool prev_base_48k = (prev_rate % 48000 == 0); + if (cur_base_48k != prev_base_48k) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } } validation: -- 2.35.3

1 0

[PATCH v3 00/15] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 31 May '22

31 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. Changes in v3: - addressed (hopefully) trace-code compilation under .configs with CONFIG_FTRACE dropped Changes in v2: - usage of avs_releast_last_firmware() dropped in error path for library loading procedure as suggested by Pierre - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre. Existing code addressed with new patch: "ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask" - fixed possible race during recovery flow (->recovering flag is now atomic and tested in single location only). - dropped platform prefixes for basically all i2s board descriptors - 'ssp_test' renamed to 'i2s_test' to match naming convention of other boards - simplified PM implementation for current series, 'low_power' bits moved to future series - replaced SND_INTEL_DSP_DRIVER_SST with _AVS as suggested by Mark. Required changes to intel-dspcfg will be added in future series - number of typos across commit messages addressed [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask include/sound/intel-dsp-config.h | 3 +- include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 501 +++++++++++ sound/soc/intel/avs/core.c | 631 +++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 +++++- sound/soc/intel/avs/loader.c | 84 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 14 +- sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 154 ++++ sound/soc/intel/avs/utils.c | 23 + 20 files changed, 3443 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

4 31

[PATCH 0/2] ASoC: da7219: Small fixes for jack detection and removal
by Adam Thomson 31 May '22

31 May '22

This series contains 2 small fixes around the AAD part of DA7219, particularly in relation to jack pole detection on certain active headsets, and tidy up when a jack is removed. Adam Thomson (2): ASoC: da7219: Fix pole orientation detection on certain headsets ASoC: da7219: cancel AAD related work earlier for jack removal sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) -- 2.17.1

2 3

[PATCH] ASoC: da7219: Drop no-op remove function
by Uwe Kleine-König 30 May '22

30 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/da7219.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c7493549a9a5..7df18a2c36b9 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -2694,11 +2694,6 @@ static int da7219_i2c_probe(struct i2c_client *i2c, return ret; } -static int da7219_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da7219_i2c_id[] = { { "da7219", }, { } @@ -2712,7 +2707,6 @@ static struct i2c_driver da7219_i2c_driver = { .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe = da7219_i2c_probe, - .remove = da7219_i2c_remove, .id_table = da7219_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

3 4

[PATCH AUTOSEL 5.18 089/159] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 7e1826d6f06f..32e6fa7b0a06 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

3 2

[PATCH AUTOSEL 4.9 20/24] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1ac96ef9ee20..703b26ec4e15 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3878,6 +3878,12 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 14/24] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 878a4fc97f04..40bf50cd87bc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3165,7 +3165,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect); @@ -3270,7 +3269,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.9 07/24] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index 36cfe1c54109..d2f9a92453f2 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 24/29] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index a98647ac497c..01de25813c72 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3972,9 +3972,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 16/29] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd3053c243c1..320d262c16c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3282,7 +3282,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3388,7 +3387,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 07/29] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index 36cfe1c54109..d2f9a92453f2 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 32/38] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9185bd7c5a6d..d34000182f67 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4105,9 +4105,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 23/38] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index ff85a0bf6170..00a90ccd6566 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3129,18 +3129,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3338,6 +3337,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 21/38] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index af9f28dd957d..4d70e6bc2c12 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3302,7 +3302,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3408,7 +3407,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 09/38] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb553cf..4742f842b457 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -77,6 +77,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index acb531749efb..074b15fcb0ac 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -48,8 +48,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -58,6 +61,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -96,8 +100,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -122,6 +129,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -242,9 +250,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -262,8 +272,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -303,10 +313,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -354,6 +368,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -373,8 +389,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -393,6 +412,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 50/55] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 466dc67799f4..dfc536cd9d2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -759,7 +759,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 46/55] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c83f7f5da96b..a66e93a3af74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4074,9 +4074,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 33/55] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index c3587af9985c..3d981441b8d1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3128,18 +3128,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3337,6 +3336,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 29/55] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1c09dfb0c0f0..56c9c4189f26 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3421,7 +3421,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3527,7 +3526,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/55] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 9eb2b5ec1ec4..78f3619f3de9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index b00ae6f39f05..e7ac82d46821 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,8 +34,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -44,6 +47,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -82,8 +86,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -108,6 +115,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -228,9 +236,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -248,8 +258,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -289,10 +299,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -340,6 +354,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -359,8 +375,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -379,6 +398,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 66/76] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 612fd7516666..25f331551f68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1098,7 +1098,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1305 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 65/76] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52c89a6f54e9..612fd7516666 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -857,7 +857,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 61/76] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 420003d062c7..d1533e95a74f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4095,9 +4095,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

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[PATCH AUTOSEL 5.10 45/76] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index d0af16b4db2f..a6f339bb4771 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3115,18 +3115,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3321,6 +3320,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 40/76] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 417732bdf286..f2f7f2dde93c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3427,7 +3427,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3529,7 +3528,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 32/76] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 43ee3d095a1b..3020a993f6ef 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -615,6 +615,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 16/76] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 9eb2b5ec1ec4..78f3619f3de9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index dc2e06ae2414..45e28db6ea38 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,8 +34,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -44,6 +47,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -82,8 +86,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -108,6 +115,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -228,9 +236,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -248,8 +258,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -289,10 +299,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -340,6 +354,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -359,8 +375,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -379,6 +398,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 094/109] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2cbf4fc0f675..d59a7e99ce42 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1148,7 +1148,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 093/109] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 47e675e8bd00..2cbf4fc0f675 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -900,7 +900,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 086/109] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..8ea6d4333562 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 062/109] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 43220bb36701..c27ca9a273e1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 054/109] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0b166e074457..47b85ba5b7d6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3428,7 +3428,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3530,7 +3529,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 045/109] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a6e837290c7d..f9c82ebc552c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -759,6 +759,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 030/109] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 029/109] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 020/109] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 119/135] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a8c6c2bfd5a7..3496403004ac 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1202,7 +1202,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 118/135] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 30c00380499c..a8c6c2bfd5a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -943,7 +943,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 106/135] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..4b2e027c1033 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 084/135] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Sasha Levin 30 May '22

30 May '22

From: Zheng Bin <zhengbin13(a)huawei.com> [ Upstream commit cbcab8cd737c74c20195c31d647e19f7cb49c9b8 ] acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> Link: https://lore.kernel.org/r/20220512013728.4128903-1-zhengbin13@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 392ffbdf6417..d809d151a38c 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -93,6 +93,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 074/135] ASoC: tscs454: Add endianness flag in snd_soc_component_driver
by Sasha Levin 30 May '22

30 May '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit ff69ec96b87dccb3a29edef8cec5d4fefbbc2055 ] The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. A fixup is also required to use the width directly rather than relying on the format in hw_params, now both little and big endian would be supported. It is worth noting this changes the behaviour of S24_LE to use a word length of 24 rather than 32. This would appear to be a correction since the fact S24_LE is stored as 32 bits should not be presented over the bus. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220504170905.332415-26-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tscs454.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 43220bb36701..c27ca9a273e1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3120,18 +3120,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3326,6 +3325,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 065/135] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb43b331a36e..f62dd119639d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3430,7 +3430,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3532,7 +3531,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 053/135] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2ace32c03ec9..b5ac226c59e1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -773,6 +773,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 037/135] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 036/135] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.17 024/135] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 141/159] ASoC: rt1015p: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b390c25c6757b9d56cecdfbf6d55f15fc89a6386 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for rt1015p, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_RT5682_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && (SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=y]) || SND_SOC_SOF_BAYTRAIL [=n] && (X86_INTEL_LPSS [=n] || COMPILE_TEST [=y])) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-3-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 156f2519459d..3dea20b2c405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1212,7 +1212,6 @@ config SND_SOC_RT1015 config SND_SOC_RT1015P tristate - depends on GPIOLIB config SND_SOC_RT1019 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 140/159] ASoC: max98357a: remove dependency on GPIOLIB
by Sasha Levin 30 May '22

30 May '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 21ca3274333f5c1cbbf9d91e5b33f4f2463859b2 ] commit dcc2c012c7691 ("ASoC: Fix gpiolib dependencies") removed a series of unnecessary dependencies on GPIOLIB when the gpio was optional. A similar simplification seems valid for max98357a, so remove the dependency as well. This will avoid the following warning WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_INTEL_SOF_CS42L42_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && (SND_SOC_SOF_HDA_LINK [=y] || SND_SOC_SOF_BAYTRAIL [=n]) && I2C [=y] && ACPI [=y] && SND_HDA_CODEC_HDMI [=y] && SND_SOC_SOF_HDA_AUDIO_CODEC [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) Reported-by: kernel test robot <yujie.liu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220517172647.468244-2-pierre-louis.bossart@linu… 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 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f46a22660103..156f2519459d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -953,7 +953,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 127/159] ASoC: rt5645: Fix errorenous cleanup order
by Sasha Levin 30 May '22

30 May '22

From: Lin Ma <linma(a)zju.edu.cn> [ Upstream commit 2def44d3aec59e38d2701c568d65540783f90f2f ] There is a logic error when removing rt5645 device as the function rt5645_i2c_remove() first cancel the &rt5645->jack_detect_work and delete the &rt5645->btn_check_timer latter. However, since the timer handler rt5645_btn_check_callback() will re-queue the jack_detect_work, this cleanup order is buggy. That is, once the del_timer_sync in rt5645_i2c_remove is concurrently run with the rt5645_btn_check_callback, the canceled jack_detect_work will be rescheduled again, leading to possible use-after-free. This patch fix the issue by placing the del_timer_sync function before the cancel_delayed_work_sync. Signed-off-by: Lin Ma <linma(a)zju.edu.cn> Link: https://lore.kernel.org/r/20220516092035.28283-1-linma@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 197c56047947..4b2e027c1033 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4154,9 +4154,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 102/159] ASoC: SOF: amd: add missing platform_device_unregister in acp_pci_rn_probe
by Sasha Levin 30 May '22

30 May '22

From: Zheng Bin <zhengbin13(a)huawei.com> [ Upstream commit cbcab8cd737c74c20195c31d647e19f7cb49c9b8 ] acp_pci_rn_probe misses a call platform_device_unregister in error path, this patch fixes that. Signed-off-by: Zheng Bin <zhengbin13(a)huawei.com> Link: https://lore.kernel.org/r/20220512013728.4128903-1-zhengbin13@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/pci-rn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 392ffbdf6417..d809d151a38c 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -93,6 +93,7 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci res = devm_kzalloc(&pci->dev, sizeof(struct resource) * ARRAY_SIZE(renoir_res), GFP_KERNEL); if (!res) { sof_pci_remove(pci); + platform_device_unregister(dmic_dev); return -ENOMEM; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 091/159] ASoC: Intel: sof_ssp_amp: fix no DMIC BE Link on Chromebooks
by Sasha Levin 30 May '22

30 May '22

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit d1c808765deb2bcd35d827402ed4d75d068aae18 ] The SOF topology supports 2 BE Links(dmic01 and dmic16k) and each link supports up to four DMICs. However, Chromebook does not implement ACPI NHLT table so the mach->mach_params.dmic_num is always zero. We add a quirk so machine driver knows it's running on a Chromebook and need to create BE Links for DMIC. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220509170922.54868-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_ssp_amp.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 88530e9de543..ef70c6f27fe1 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -9,6 +9,7 @@ #include <linux/acpi.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -78,6 +79,16 @@ struct sof_card_private { bool idisp_codec; }; +static const struct dmi_system_id chromebook_platforms[] = { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {}, +}; + static const struct snd_soc_dapm_widget sof_ssp_amp_dapm_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; @@ -371,7 +382,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_links; struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; - int dmic_be_num, hdmi_num = 0; + int dmic_be_num = 0, hdmi_num = 0; int ret, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); @@ -383,7 +394,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) mach = pdev->dev.platform_data; - dmic_be_num = mach->mach_params.dmic_num; + if (dmi_check_system(chromebook_platforms) || mach->mach_params.dmic_num > 0) + dmic_be_num = 2; ssp_codec = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 079/159] ASoC: SOF: ipc3-topology: Correct get_control_data for non bytes payload
by Sasha Levin 30 May '22

30 May '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit a962890a5a3cce903ff7c7a19fadee63ed9efdc7 ] It is possible to craft a topology where sof_get_control_data() would do out of bounds access because it expects that it is only called when the payload is bytes type. Confusingly it also handles other types of controls, but the payload parsing implementation is only valid for bytes. Fix the code to count the non bytes controls and instead of storing a pointer to sof_abi_hdr in sof_widget_data (which is only valid for bytes), store the pointer to the data itself and add a new member to save the size of the data. In case of non bytes controls we store the pointer to the chanv itself, which is just an array of values at the end. In case of bytes control, drop the wrong cdata->data (wdata[i].pdata) check against NULL since it is incorrect and invalid in this context. The data is pointing to the end of cdata struct, so it should never be null. Reported-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Tested-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Link: https://lore.kernel.org/r/20220427185221.28928-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/ipc3-topology.c | 39 +++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 2f8450a8c0a1..af1bbd34213c 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,7 +20,8 @@ struct sof_widget_data { int ctrl_type; int ipc_cmd; - struct sof_abi_hdr *pdata; + void *pdata; + size_t pdata_size; struct snd_sof_control *control; }; @@ -784,16 +785,26 @@ static int sof_get_control_data(struct snd_soc_component *scomp, } cdata = wdata[i].control->ipc_control_data; - wdata[i].pdata = cdata->data; - if (!wdata[i].pdata) - return -EINVAL; - /* make sure data is valid - data can be updated at runtime */ - if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && - wdata[i].pdata->magic != SOF_ABI_MAGIC) - return -EINVAL; + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES) { + /* make sure data is valid - data can be updated at runtime */ + if (cdata->data->magic != SOF_ABI_MAGIC) + return -EINVAL; + + wdata[i].pdata = cdata->data->data; + wdata[i].pdata_size = cdata->data->size; + } else { + /* points to the control data union */ + wdata[i].pdata = cdata->chanv; + /* + * wdata[i].control->size is calculated with struct_size + * and includes the size of struct sof_ipc_ctrl_data + */ + wdata[i].pdata_size = wdata[i].control->size - + sizeof(struct sof_ipc_ctrl_data); + } - *size += wdata[i].pdata->size; + *size += wdata[i].pdata_size; /* get data type */ switch (cdata->cmd) { @@ -876,10 +887,12 @@ static int sof_process_load(struct snd_soc_component *scomp, */ if (ipc_data_size) { for (i = 0; i < widget->num_kcontrols; i++) { - memcpy(&process->data[offset], - wdata[i].pdata->data, - wdata[i].pdata->size); - offset += wdata[i].pdata->size; + if (!wdata[i].pdata_size) + continue; + + memcpy(&process->data[offset], wdata[i].pdata, + wdata[i].pdata_size); + offset += wdata[i].pdata_size; } } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 078/159] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ca917a849c42..869c76506b66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3437,7 +3437,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3539,7 +3538,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 064/159] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220427134918.527381-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d76a505052fb..f81ae742faa7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -773,6 +773,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 045/159] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87pmlbgn5t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..af8ef2a27d34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 044/159] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87r15rgn6p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 028/159] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

1 0

[PATCH] ALSA: hda/realtek - Fix microphone noise on ASUS TUF B550M-PLUS
by Marios Levogiannis 30 May '22

30 May '22

Set microphone pins 0x18 (rear) and 0x19 (front) to VREF_50 to fix the microphone noise on ASUS TUF B550M-PLUS which uses the ALCS1200A codec. The initial value was VREF_80. The same issue is also present on Windows using both the default Windows driver and all tested Realtek drivers before version 6.0.9049.1. Comparing Realtek driver 6.0.9049.1 (the first one without the microphone noise) to Realtek driver 6.0.9047.1 (the last one with the microphone noise) revealed that the fix is the result of setting pins 0x18 and 0x19 to VREF_50. This fix may also work for other boards that have been reported to have the same microphone issue and use the ALC1150 and ALCS1200A codecs, since these codecs are similar and the fix in the Realtek driver on Windows is common for both. However, it is currently enabled only for ASUS TUF B550M-PLUS as this is the only board that could be tested. --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 89d610afa..f3ad454b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1981,6 +1981,7 @@ enum { ALC1220_FIXUP_CLEVO_PB51ED_PINS, ALC887_FIXUP_ASUS_AUDIO, ALC887_FIXUP_ASUS_HMIC, + ALCS1200A_FIXUP_MIC_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2526,6 +2527,14 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC887_FIXUP_ASUS_AUDIO, }, + [ALCS1200A_FIXUP_MIC_VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF50 }, /* rear mic */ + { 0x19, PIN_VREF50 }, /* front mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2563,6 +2572,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1043, 0x8797, "ASUS TUF B550M-PLUS", ALCS1200A_FIXUP_MIC_VREF), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), -- 2.36.1

2 3

[PATCH 00/15] rewrite public API according to convention of GNOME project
by GitHub pull_request - opened 30 May '22

30 May '22

alsa-project/alsa-gobject pull request #70 was opened from takaswie: In GNOME convention for throw function which has an argument of GError to report failure, the function is programmed to return value of gboolean type to report whether the argument is filled with instance of GError. On the other hand, I wrote the throw functions to return nothing (void). This patchset rewrite public API according to it. Unfortunately, it brings loss of backward compatibility. ``` Takashi Sakamoto (15): ctl: remove useless exposed symbols timer: user-instance: fix closure marshaller for event signal ctl: query: rewrite public API to return gboolean according to GNOME convention ctl: card: rewrite public API to return gboolean according to GNOME convention ctl: elem_info: rewrite public API to return gboolean according to GNOME convention rawmidi: query: rewrite public API to return gboolean according to GNOME convention rawmidi: stream_pair: rewrite public API to return gboolean according to GNOME convention hwdep: query: rewrite public API to return gboolean according to GNOME convention timer: query: rewrite public API to return gboolean according to GNOME convention timer: user_instance: rewrite public API to return gboolean according to GNOME convention timer: instance-params: rewrite public API to return gboolean according to GNOME convention seq: query: rewrite public API to return gboolean according to GNOME convention seq: user_client: rewrite public API to return gboolean according to GNOME convention seq: client_info: rewrite public API to return gboolean according to GNOME convention seq: event_cntr: rewrite public API to return gboolean according to GNOME convention samples/ctl | 21 +- samples/hwdep | 8 +- samples/rawmidi | 14 +- samples/seq | 37 +- samples/timer | 9 +- src/ctl/alsactl.map | 73 ++-- src/ctl/card.c | 322 +++++++++------- src/ctl/card.h | 82 ++--- src/ctl/elem-info.c | 84 +++-- src/ctl/elem-info.h | 12 +- src/ctl/query.c | 59 ++- src/ctl/query.h | 8 +- src/hwdep/alsahwdep.map | 13 +- src/hwdep/query.c | 64 ++-- src/hwdep/query.h | 17 +- src/rawmidi/alsarawmidi.map | 37 +- src/rawmidi/privates.h | 2 +- src/rawmidi/query.c | 96 +++-- src/rawmidi/query.h | 37 +- src/rawmidi/stream-pair.c | 163 +++++---- src/rawmidi/stream-pair.h | 31 +- src/seq/alsaseq.map | 98 ++--- src/seq/client-info.c | 38 +- src/seq/client-info.h | 16 +- src/seq/event-cntr.c | 711 +++++++++++++++++++++--------------- src/seq/event-cntr.h | 203 +++++----- src/seq/query.c | 298 +++++++++------ src/seq/query.h | 46 +-- src/seq/user-client.c | 413 +++++++++++++-------- src/seq/user-client.h | 140 +++---- src/timer/alsatimer.map | 44 ++- src/timer/instance-params.c | 30 +- src/timer/instance-params.h | 4 +- src/timer/query.c | 163 ++++++--- src/timer/query.h | 23 +- src/timer/user-instance.c | 192 ++++++---- src/timer/user-instance.h | 43 +-- 37 files changed, 2117 insertions(+), 1534 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/70 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v2 0/6] OPE support on Tegra210 and later
by Sameer Pujar 30 May '22

30 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

2 9

ucm2 - splitting 10 channel interface into mix of mono and stereo channels
by GitHub issues - opened 29 May '22

29 May '22

alsa-project/alsa-ucm-conf issue #161 was opened from smangels: Hi, I've pushed a PR for the Behringer Flow8 digital mixer configured to be in "streaming" mode. Now that I've managed to wrap my head around the configuration, I would like to continue with the so called "recording" mode (different ID in `lsusb`). Thus I wonder whether UCM2 syntax allows me to split a 10 channel interface into a mix of mono and stereo channels (mic - mono, master, stereo)? Regards, Sebastian Mangelsen Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/161 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties
by Nícolas F. R. A. Prado 29 May '22

29 May '22

The Mediatek AFE PCM controller for MT8192 allows two I2S interfaces to share the same clock and act as a single interface with both input and output. Add patterns for these properties in the dt-binding. The property is split into two patterns in order to allow all valid interface pairings. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- The series from v1 of this patch was merged although some changes were still needed in this patch, so the v1 of this patch was reverted [1] and this standalone commit addresses the feedback from v1 and readds the property. [1] https://lore.kernel.org/all/20220509185625.580811-1-nfraprado@collabora.com v1: https://lore.kernel.org/all/20220429203039.2207848-2-nfraprado@collabora.co… Changes in v2: - Added "mediatek," prefix to property - Rewrote and added more information to property description - Split into two patterns to validate that output-input pairings are done .../bindings/sound/mt8192-afe-pcm.yaml | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml index 7a25bc9b8060..2abf43c6c2c3 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -54,6 +54,22 @@ properties: - const: aud_infra_clk - const: aud_infra_26m_clk +patternProperties: + "^mediatek,i2s[13579]-share$": + description: + Each I2S interface has a single data line, input if its index is even or + output if the index is odd. An input and an output I2S interface can be + used together as if they were a single I2S interface with both input and + output data lines by sharing the same clock. This property represents this + pairing. The value should be the name of the interface whose clock is + used, and the property name the other interface that depends on this + clock. + pattern: "^I2S[0268]$" + + "^mediatek,i2s[0268]-share$": + description: Same as above. + pattern: "^I2S[13579]$" + required: - compatible - interrupts -- 2.36.0

3 5

ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 29 May '22

29 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input, when vendor ID product ID are `1397:050d` - prepare filenames for Flow8's recording mode `1397:050c` (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 29 May '22

29 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input - prepare filenames for Flow8's recording mode (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - opened 29 May '22

29 May '22

alsa-project/alsa-ucm-conf pull request #160 was opened from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow configured into "streaming" mode - 2 stereo outputs, 1 stereo input - prepare filenames for Flow8's recording mode (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Behringer Flow8
by GitHub pull_request - opened 28 May '22

28 May '22

alsa-project/alsa-ucm-conf pull request #159 was opened from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow configured into "streaming" mode - 2 stereo outputs, 1 stereo input Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/159 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/159.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/via: Delete does not require return
by Xiang wangx 27 May '22

27 May '22

Void function return statements are not generally useful. Signed-off-by: Xiang wangx <wangxiang(a)cdjrlc.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 773a136161f1..a05304f340df 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -449,8 +449,6 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); snd_hda_codec_set_pincfg(codec, nid, def_conf); } - - return; } static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol, -- 2.36.1

2 1

[Patch] ALSA: hda/realtek: Enable 4-speaker output for Dell XPS 15 9520 laptop
by Rik van der Kemp 27 May '22

27 May '22

The 2022-model XPS 15 appears to use the same 4-speakers-on-ALC289 audio setup as the Dell XPS 15 9510, so requires the same quirk to enable woofer output. Tested on my own 9520. Signed-off-by: Rik van der Kemp <rik(a)upto11.nl> --- 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 80e4955e8c10..e64597f2675b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8902,6 +8902,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216035 Signed-off-by: Rik van der Kemp <rik(a)upto11.nl>

2 1

[PATCH] Enable SPDIF output on Intel Hades Canyon
by Dag B 27 May '22

27 May '22

Without the attached patch, the s/pdif output on the Hades Canyon NUC does not work. "Well known" issue, less known fix. As far as I can tell, there is no risk of any averse side-effects. But a bonus fix is enabling on-wire headset microphone, by chaining the right 'model' choice for the hardware. If I should Cc: someone directly for this patch to be picked up, please let me know. I have been running with this patch for half a year or so. Patch passes checkpatch.pl Patch is based on what 'cyber4o' posted on the insanelymac forum [1]. All the glory to this person, any error is likely mine. Some other users with issues: [2] [3] Dag Bakke [1] https://www.insanelymac.com/forum/topic/339291-guide-hac-mini-osx-mojave-on… 2] https://bbs.archlinux.org/viewtopic.php?id=270917 [3] https://www.reddit.com/r/intelnuc/comments/9ft9x8/any_linux_users_got_the_s…

3 6

[PATCH] ASoC: da732x: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/da732x.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 42d6a3fc3af5..267936015cb8 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1547,11 +1547,6 @@ static int da732x_i2c_probe(struct i2c_client *i2c, return ret; } -static int da732x_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id da732x_i2c_id[] = { { "da7320", 0}, { } @@ -1563,7 +1558,6 @@ static struct i2c_driver da732x_i2c_driver = { .name = "da7320", }, .probe = da732x_i2c_probe, - .remove = da732x_i2c_remove, .id_table = da732x_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

2 1

[PATCH] ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe
by zhangqilong 27 May '22

27 May '22

Add missing pm_runtime_disable() when probe error out. It could avoid pm_runtime implementation complains when removing and probing again the driver. Fix:c590fa80b3928 ("ASoC: fsl_xcvr: register platform component before registering cpu dai") Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/soc/fsl/fsl_xcvr.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index d0556c79fdb1..907435efe121 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1228,6 +1228,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) */ ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to pcm register\n"); return ret; } @@ -1235,6 +1236,7 @@ static int fsl_xcvr_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp, &fsl_xcvr_dai, 1); if (ret) { + pm_runtime_disable(dev); dev_err(dev, "failed to register component %s\n", fsl_xcvr_comp.name); } -- 2.31.1

2 1

[PATCH] ASoC: lm49453: Drop no-op remove function
by Uwe Kleine-König 27 May '22

27 May '22

A remove callback that just returns 0 is equivalent to no callback at all as can be seen in i2c_device_remove(). So simplify accordingly. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/lm49453.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 973d781b4b6a..86cead5022c0 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1443,11 +1443,6 @@ static int lm49453_i2c_probe(struct i2c_client *i2c, return ret; } -static int lm49453_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id lm49453_i2c_id[] = { { "lm49453", 0 }, { } @@ -1459,7 +1454,6 @@ static struct i2c_driver lm49453_i2c_driver = { .name = "lm49453", }, .probe = lm49453_i2c_probe, - .remove = lm49453_i2c_remove, .id_table = lm49453_i2c_id, }; base-commit: 4b0986a3613c92f4ec1bdc7f60ec66fea135991f -- 2.36.1

1 1

[PATCH] ASoC: Intel: common: fix typo for tplg naming
by David Lin 26 May '22

26 May '22

Correct typo form sof-adl-mx98360a-nau8825.tplg to sof-adl-max98360a-nau8825.tplg. The reason is tplg naming without naming limitaion of length. It will be consistency with sof topology generation. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index e6e52c7b6803..c1385161cdc8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -453,7 +453,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .drv_name = "adl_mx98360a_nau8825", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", }, { .id = "RTL5682", -- 2.25.1

3 2

[PATCH] ASoC: rt5640: Do not manipulate pin "Platform Clock" if the "Platform Clock" is not in the DAPM
by Oder Chiou 26 May '22

26 May '22

The pin "Platform Clock" was only used by the Intel Byt CR platform. In the others, the error log will be informed. The patch will set the flag to avoid the pin "Platform Clock" manipulated by the other platforms. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 11 +++++++++-- sound/soc/codecs/rt5640.h | 2 ++ sound/soc/intel/boards/bytcr_rt5640.c | 2 ++ 3 files changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 12da2bea1a7b..69c80d80ed9d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2094,12 +2094,14 @@ EXPORT_SYMBOL_GPL(rt5640_sel_asrc_clk_src); void rt5640_enable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); /* OVCD is unreliable when used with RCCLK as sysclk-source */ - snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_sync_unlocked(dapm); snd_soc_dapm_mutex_unlock(dapm); } @@ -2108,9 +2110,11 @@ EXPORT_SYMBOL_GPL(rt5640_enable_micbias1_for_ovcd); void rt5640_disable_micbias1_for_ovcd(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); snd_soc_dapm_mutex_lock(dapm); - snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); + if (rt5640->use_platform_clock) + snd_soc_dapm_disable_pin_unlocked(dapm, "Platform Clock"); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); snd_soc_dapm_disable_pin_unlocked(dapm, "LDO2"); snd_soc_dapm_sync_unlocked(dapm); @@ -2535,6 +2539,9 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, rt5640->jd_gpio_irq_requested = true; } + if (jack_data && jack_data->use_platform_clock) + rt5640->use_platform_clock = jack_data->use_platform_clock; + ret = request_irq(rt5640->irq, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 9e49b9a0ccaa..505c93514051 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2155,11 +2155,13 @@ struct rt5640_priv { bool jd_inverted; unsigned int ovcd_th; unsigned int ovcd_sf; + bool use_platform_clock; }; struct rt5640_set_jack_data { int codec_irq_override; struct gpio_desc *jd_gpio; + bool use_platform_clock; }; int rt5640_dmic_enable(struct snd_soc_component *component, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7b948a219177..ed9fa1728722 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1191,12 +1191,14 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct byt_rt5640_private *priv = snd_soc_card_get_drvdata(card); + struct rt5640_set_jack_data *jack_data = &priv->jack_data; struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; const struct snd_soc_dapm_route *custom_map = NULL; int num_routes = 0; int ret; card->dapm.idle_bias_off = true; + jack_data->use_platform_clock = true; /* Start with RC clk for jack-detect (we disable MCLK below) */ if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) -- 2.35.3

3 2

Update mapping for the new Gigabyte Aorus Pro WiFi rev 1.2
by GitHub pull_request - edited 26 May '22

26 May '22

alsa-project/alsa-ucm-conf pull request #156 was edited from 13r0ck: Requires kernel patch from: https://bugzilla.kernel.org/attachment.cgi?id=301036&action=diff This doesn't fix the front mic for some reason (any ideas please?)~~, as well I am not sure if this will cause a regression for the 1.1 board~~ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/156.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Cancel pending work at closing a MIDI substream
by Takashi Iwai 25 May '22

25 May '22

At closing a USB MIDI output substream, there might be still a pending work, which would eventually access the rawmidi runtime object that is being released. For fixing the race, make sure to cancel the pending work at closing. Reported-by: syzbot+6912c9592caca7ca0e7d(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000e7e75005dfd07cf6@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7c6ca2b433a5..344fbeadf161 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1145,6 +1145,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct usbmidi_out_port *port = substream->runtime->private_data; + + cancel_work_sync(&port->ep->work); return substream_open(substream, 0, 0); } -- 2.35.3

1 0