October 2021 - Alsa-devel - mailman.alsa-project.org

Mute LED on HP OMEN 15 needs a fixup in hda_intel
by Влад Макаров 24 Oct '21

24 Oct '21

Greetings everyone I am not sure if this is the right place to ask, but I don't know where else to look for help. I have a laptop (HP Omen 15 2020 amd) which has a mute button (Fn + F5) which works great, however, the LED indicator on it doesn't light up. With the help of this article <https://asus-linux.org/blog/sound-2021-01-11/> I managed to figure out that I can trigger the LED with the following command: sudo hda-verb /dev/snd/hwC1D0 0x20 0x500 0x0B && sudo hda-verb /dev/snd/hwC1D0 0x20 0x400 0x7778 And disable it with sudo hda-verb /dev/snd/hwC1D0 0x20 0x500 0x0B && sudo hda-verb /dev/snd/hwC1D0 0x20 0x400 0x777 I think that a fix should be added to patch_realtek.c <https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/tree/sound…> (to change those values when mute button is pressed) but this requires knowledge of C and kernel stuff. I don't have much experience with those so I would be very grateful if anyone could help me. I have also filed a kernel bug report <https://bugzilla.kernel.org/show_bug.cgi?id=214735> (with some useful attachments) but nobody responded there. Thanks in advance

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[syzbot] possible deadlock in snd_mixer_oss_ioctl1
by syzbot 24 Oct '21

24 Oct '21

Hello, syzbot found the following issue on: HEAD commit: cf6c9d12750c Add linux-next specific files for 20211022 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=15d3d4c4b00000 kernel config: https://syzkaller.appspot.com/x/.config?x=e694204255ade3a3 dashboard link: https://syzkaller.appspot.com/bug?extid=ace149a75a9a0a399ac7 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=17c0530cb00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1129c952b00000 The issue was bisected to: commit 411cef6adfb38a5bb6bd9af3941b28198e7fb680 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Oct 20 16:48:46 2021 +0000 ALSA: mixer: oss: Fix racy access to slots bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=167f6454b00000 final oops: https://syzkaller.appspot.com/x/report.txt?x=157f6454b00000 console output: https://syzkaller.appspot.com/x/log.txt?x=117f6454b00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+ace149a75a9a0a399ac7(a)syzkaller.appspotmail.com Fixes: 411cef6adfb3 ("ALSA: mixer: oss: Fix racy access to slots") ============================================ WARNING: possible recursive locking detected 5.15.0-rc6-next-20211022-syzkaller #0 Not tainted -------------------------------------------- syz-executor206/6529 is trying to acquire lock: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:316 [inline] ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x630/0x19a0 sound/core/oss/mixer_oss.c:375 but task is already holding lock: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:300 [inline] ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x4b1/0x19a0 sound/core/oss/mixer_oss.c:375 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(&mixer->reg_mutex); lock(&mixer->reg_mutex); *** DEADLOCK *** May be due to missing lock nesting notation 1 lock held by syz-executor206/6529: #0: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:300 [inline] #0: ffff888021ff09c8 (&mixer->reg_mutex){+.+.}-{3:3}, at: snd_mixer_oss_ioctl1+0x4b1/0x19a0 sound/core/oss/mixer_oss.c:375 stack backtrace: CPU: 0 PID: 6529 Comm: syz-executor206 Not tainted 5.15.0-rc6-next-20211022-syzkaller #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_deadlock_bug kernel/locking/lockdep.c:2956 [inline] check_deadlock kernel/locking/lockdep.c:2999 [inline] validate_chain kernel/locking/lockdep.c:3788 [inline] __lock_acquire.cold+0x149/0x3ab kernel/locking/lockdep.c:5027 lock_acquire kernel/locking/lockdep.c:5637 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5602 __mutex_lock_common kernel/locking/mutex.c:607 [inline] __mutex_lock+0x12f/0x12f0 kernel/locking/mutex.c:740 snd_mixer_oss_set_volume sound/core/oss/mixer_oss.c:316 [inline] snd_mixer_oss_ioctl1+0x630/0x19a0 sound/core/oss/mixer_oss.c:375 snd_mixer_oss_ioctl+0x40/0x50 sound/core/oss/mixer_oss.c:390 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 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+0x44/0xae RIP: 0033:0x7f0c06979fe9 Code: 28 c3 e8 2a 14 00 00 66 2e 0f 1f 84 00 00 00 00 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 c0 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007ffea97dd558 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 00007f0c06979fe9 RDX: 0000000020000080 RSI: 00000000c0044d00 RDI: 0000000000000003 RBP: 00007f0c0693dfd0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f0c0693e060 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 </TASK> --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

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Where did snd_pcm_parse_control_id() go?
by GitHub issues - opened 24 Oct '21

24 Oct '21

alsa-project/alsa-lib issue #186 was opened from Saur2000: It seems `snd_pcm_parse_control_id()` was made private in 1.2.5 without any prior deprecation. It is still present in `pcm_external.h`, and our code now fails to link with: ``` ld: src/plugin_control.o: in function `plugin_control_load_user_control': src/plugin_control.c:254: undefined reference to `snd_pcm_parse_control_id' ``` What are we supposed to use instead? And why is the function still in the header file if it is not in the library? Issue URL : https://github.com/alsa-project/alsa-lib/issues/186 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] sound/usb: fix null pointer dereference on pointer cs_desc
by Chengfeng Ye 24 Oct '21

24 Oct '21

The pointer cs_desc return from snd_usb_find_clock_source could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Fixes: 9ec73005 ("ALSA: usb-audio: Refactoring UAC2/3 clock setup code") Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/usb/clock.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548..98345a695dcc 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -496,6 +496,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, union uac23_clock_source_desc *cs_desc; cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); + + if (!cs_desc) + return 0; + if (fmt->protocol == UAC_VERSION_3) bmControls = le32_to_cpu(cs_desc->v3.bmControls); else -- 2.17.1

1 0

[PATCH V2 00/13] Add Yellow Carp platform ASoC driver
by Vijendar Mukunda 23 Oct '21

23 Oct '21

Yellow Carp platform is new APU series based on acp6.x design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Yellow Carp APU with DMIC endpoint support. changes since v1: - fixed SPDX commenting style in acp6x pci driver. - fixed Copyright commenting style in acp6x pci driver, pdm dma driver and machine driver files. - fixed local variable check logic in acp6x_power_on() function. - fixed Kconfig indentation issue - removed irq_flag local variable in irq handler. - add dmi info check in machine driver probe Vijendar Mukunda (13): ASoC: amd: add Yellow Carp ACP6x IP register header ASoC: amd: add Yellow Carp ACP PCI driver ASoC: amd: add acp6x init/de-init functions ASoC: amd: add platform devices for acp6x pdm driver and dmic driver ASoC: amd: add acp6x pdm platform driver ASoC: amd: add acp6x irq handler ASoC: amd: add acp6x pdm driver dma ops ASoC: amd: add acp6x pci driver pm ops ASoC: amd: add acp6x pdm driver pm ops ASoC: amd: enable Yellow carp acp6x drivers build ASoC: amd: create platform device for acp6x machine driver ASoC: amd: add YC machine driver using dmic ASoC: amd: enable Yellow Carp platform machine driver build sound/soc/amd/Kconfig | 21 + sound/soc/amd/Makefile | 1 + sound/soc/amd/yc/Makefile | 9 + sound/soc/amd/yc/acp6x-mach.c | 194 ++++++++++ sound/soc/amd/yc/acp6x-pdm-dma.c | 448 ++++++++++++++++++++++ sound/soc/amd/yc/acp6x.h | 107 ++++++ sound/soc/amd/yc/acp6x_chip_offset_byte.h | 444 +++++++++++++++++++++ sound/soc/amd/yc/pci-acp6x.c | 338 ++++++++++++++++ 8 files changed, 1562 insertions(+) create mode 100644 sound/soc/amd/yc/Makefile create mode 100644 sound/soc/amd/yc/acp6x-mach.c create mode 100644 sound/soc/amd/yc/acp6x-pdm-dma.c create mode 100644 sound/soc/amd/yc/acp6x.h create mode 100644 sound/soc/amd/yc/acp6x_chip_offset_byte.h create mode 100644 sound/soc/amd/yc/pci-acp6x.c -- 2.25.1

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[PATCH v3 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 23 Oct '21

23 Oct '21

changes since v2: - Fixes Kconfig description and indent. Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 51 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 600 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1993 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

3 13

[PATCH v4 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 23 Oct '21

23 Oct '21

changes since v3: - rebase and fixes merge conflict. - Fixed kernel autobot warning. Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 51 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 340 +++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 600 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 315 ++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 131 ++++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1993 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

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[PATCH v3 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 23 Oct '21

23 Oct '21

Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/Kconfig | 13 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/common.c | 39 ++ sound/soc/qcom/common.h | 1 + sound/soc/qcom/lpass-cdc-dma.c | 195 +++++++ sound/soc/qcom/lpass-cpu.c | 245 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 513 +++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 599 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 157 ++++++ 11 files changed, 1859 insertions(+), 79 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 11

[PATCH 0/2] ASoC: meson: axg: fix TDM channel order sync
by Jerome Brunet 23 Oct '21

23 Oct '21

On the Amlogic AXG series, the TODDR FIFO may get out of sync with the TDM decoder if the decoder is started before the FIFO. The channel appears shifted in memory in an unpredictable way. To fix this, the trick is to start the FIFO before the TDM decoder. This way the FIFO is already waiting when the 1st channel is produced and it is correctly placed in memory. Jerome Brunet (2): ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++++++----- 2 files changed, 22 insertions(+), 5 deletions(-) -- 2.33.0

2 3

[PATCH] ASoC: rt5682s: Downsizing the DAC volume scale
by derek.fang＠realtek.com 23 Oct '21

23 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> Use 0.75db/step of DAC volume instead of 1.5 to get a more smooth volume curve. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 9dd1796f3339..f6435b233a51 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1003,7 +1003,7 @@ static int rt5682s_set_jack_detect(struct snd_soc_component *component, return 0; } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9562, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); @@ -1011,7 +1011,7 @@ static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, - RT5682S_L_VOL_SFT + 2, RT5682S_R_VOL_SFT + 2, 63, 0, dac_vol_tlv), + RT5682S_L_VOL_SFT + 1, RT5682S_R_VOL_SFT + 1, 127, 0, dac_vol_tlv), /* CBJ Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682S_REC_MIXER, -- 2.17.1

2 1

[PATCH] ASoC: cs42l42: Remove unused runtime_suspend/runtime_resume callbacks
by Richard Fitzgerald 23 Oct '21

23 Oct '21

The driver has runtime_suspend and runtime_resume callbacks, but pm_runtime is never enabled so these functions won't be called. They could not be used anyway because the runtime_suspend would cause jack detect to stop working. These functions are unused - delete them. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 51 +--------------------------------------------- 1 file changed, 1 insertion(+), 50 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5e4d6791756b..0dbe4e23194b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -25,7 +25,6 @@ #include <linux/regulator/consumer.h> #include <linux/gpio/consumer.h> #include <linux/of_device.h> -#include <linux/pm_runtime.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2175,59 +2174,12 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) if (i2c_client->irq) devm_free_irq(&i2c_client->dev, i2c_client->irq, cs42l42); - pm_runtime_suspend(&i2c_client->dev); - pm_runtime_disable(&i2c_client->dev); - - return 0; -} - -#ifdef CONFIG_PM -static int cs42l42_runtime_suspend(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - - regcache_cache_only(cs42l42->regmap, true); - regcache_mark_dirty(cs42l42->regmap); - - /* Hold down reset */ gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); - - /* remove power */ - regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), cs42l42->supplies); return 0; } -static int cs42l42_runtime_resume(struct device *dev) -{ - struct cs42l42_private *cs42l42 = dev_get_drvdata(dev); - int ret; - - /* Enable power */ - ret = regulator_bulk_enable(ARRAY_SIZE(cs42l42->supplies), - cs42l42->supplies); - if (ret != 0) { - dev_err(dev, "Failed to enable supplies: %d\n", - ret); - return ret; - } - - gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); - usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); - - regcache_cache_only(cs42l42->regmap, false); - regcache_sync(cs42l42->regmap); - - return 0; -} -#endif - -static const struct dev_pm_ops cs42l42_runtime_pm = { - SET_RUNTIME_PM_OPS(cs42l42_runtime_suspend, cs42l42_runtime_resume, - NULL) -}; - #ifdef CONFIG_OF static const struct of_device_id cs42l42_of_match[] = { { .compatible = "cirrus,cs42l42", }, @@ -2254,7 +2206,6 @@ MODULE_DEVICE_TABLE(i2c, cs42l42_id); static struct i2c_driver cs42l42_i2c_driver = { .driver = { .name = "cs42l42", - .pm = &cs42l42_runtime_pm, .of_match_table = of_match_ptr(cs42l42_of_match), .acpi_match_table = ACPI_PTR(cs42l42_acpi_match), }, -- 2.11.0

2 1

[PATCH] ASoC: fix unmet dependency on GPIOLIB
by Julian Braha 23 Oct '21

23 Oct '21

When SND_SOC_SC7180 or SND_SOC_STORM is selected, and GPIOLIB is not selected, Kbuild gives 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_STORM [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] - SND_SOC_SC7180 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_QCOM [=y] && I2C [=y] This is because SND_SOC_MAX98357A is selected by SND_SOC_STORM and SND_SOC_SC7180, but these config options do not select or depend on GPIOLIB, despite SND_SOC_MAX98357A depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/qcom/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index cc7c1de2f1d9..8c5c1ed54bc7 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -38,6 +38,7 @@ config SND_SOC_LPASS_SC7180 config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" + depends on GPIOLIB select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -141,7 +142,7 @@ config SND_SOC_SM8250 config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" - depends on I2C + depends on I2C && GPIOLIB select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A -- 2.25.1

2 1

wrong test in alsa-info.sh
by GitHub issues - opened 22 Oct '21

22 Oct '21

alsa-project/alsa-utils issue #115 was opened from mhampl: In line 691 of alsa_info.sh script (version 0.5.0) a test for $TEMPDIR/alsa-usbstream.tmp is needed instead of $TEMPDIR/lsusb.tmp Issue URL : https://github.com/alsa-project/alsa-utils/issues/115 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: cs42l51: improve error handling in cs42l51_remove()
by Uwe Kleine-König 22 Oct '21

22 Oct '21

When disabling a regulator fails while the device goes away, it doesn't make sense to reenable the already disabled regulators, so use regulator_bulk_force_disable() instead of regulator_bulk_disable(). In the error case returning an error code is also hardly sensible because the only effect is that the i2c core emits a generic error message. Replace this by a more specific message and make sure the i2c remove callback returns 0 instead. While at it also change cs42l51_remove() to return void as it doesn't yield any meaningful return value any more. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, a bit unsure if regulator_bulk_force_disable() is indeed the right function here, its documentation specifies a different usecase. My motivation for this change was to make it obvious in cs42l51_i2c_remove() that there is no error to handle to eventually make i2c remove callbacks return void, too. Best regards Uwe sound/soc/codecs/cs42l51-i2c.c | 4 +++- sound/soc/codecs/cs42l51.c | 11 ++++++++--- sound/soc/codecs/cs42l51.h | 2 +- 3 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 70260e0a8f09..3cb21a2ba29f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -31,7 +31,9 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, static int cs42l51_i2c_remove(struct i2c_client *i2c) { - return cs42l51_remove(&i2c->dev); + cs42l51_remove(&i2c->dev); + + return 0; } static const struct dev_pm_ops cs42l51_pm_ops = { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..f3667693e2ea 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -793,14 +793,19 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs42l51_probe); -int cs42l51_remove(struct device *dev) +void cs42l51_remove(struct device *dev) { struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + int ret; gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); - return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), - cs42l51->supplies); + ret = regulator_bulk_force_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret) + dev_warn(dev, "Failed to disable all regulators (%pe)\n", + ERR_PTR(ret)); + } EXPORT_SYMBOL_GPL(cs42l51_remove); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 9d06cf7f8876..a79343e8a54e 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -13,7 +13,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); -int cs42l51_remove(struct device *dev); +void cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; -- 2.30.2

2 5

[PATCH] ASoC: Intel: Unify HDAudio-ext bus initialization
by Cezary Rojewski 21 Oct '21

21 Oct '21

HDAudio-extended bus initialization parts are scattered throughout Intel ADSP drivers code. Gather them up in snd_hda_ext_bus_init() to provide unified initialization point. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/hdaudio_ext.h | 9 ++++++--- sound/hda/ext/hdac_ext_bus.c | 29 ++++++++++++++++++++--------- sound/soc/intel/skylake/skl.c | 9 +-------- sound/soc/sof/intel/hda-bus.c | 16 +++++++++------- sound/soc/sof/intel/hda.c | 5 +++-- sound/soc/sof/intel/hda.h | 3 ++- 6 files changed, 41 insertions(+), 30 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 375581634143..d1f6e9f7c057 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -2,11 +2,14 @@ #ifndef __SOUND_HDAUDIO_EXT_H #define __SOUND_HDAUDIO_EXT_H +#include <linux/pci.h> #include <sound/hdaudio.h> +#include <sound/hda_codec.h> -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops); +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model); void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 765c40a6ccba..a89e2e80ea4c 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -10,15 +10,17 @@ */ #include <linux/module.h> +#include <linux/pci.h> #include <linux/slab.h> #include <linux/io.h> #include <sound/hdaudio_ext.h> +#include <sound/hda_codec.h> MODULE_DESCRIPTION("HDA extended core"); MODULE_LICENSE("GPL v2"); /** - * snd_hdac_ext_bus_init - initialize a HD-audio extended bus + * snd_hda_ext_bus_init - initialize a HD-audio extended bus * @bus: the pointer to HDAC bus object * @dev: device pointer * @ops: bus verb operators @@ -26,28 +28,37 @@ MODULE_LICENSE("GPL v2"); * * Returns 0 if successful, or a negative error code. */ -int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_bus_ops *ops, - const struct hdac_ext_bus_ops *ext_ops) +int snd_hda_ext_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const struct hdac_bus_ops *ops, + const struct hdac_ext_bus_ops *ext_ops, + const char *model) { + struct hdac_bus *base = &bus->core; int ret; - ret = snd_hdac_bus_init(bus, dev, ops); + ret = snd_hdac_bus_init(base, &pdev->dev, ops); if (ret < 0) return ret; - bus->ext_ops = ext_ops; + base->ext_ops = ext_ops; /* FIXME: * Currently only one bus is supported, if there is device with more * buses, bus->idx should be greater than 0, but there needs to be a * reliable way to always assign same number. */ - bus->idx = 0; - bus->cmd_dma_state = true; + base->idx = 0; + base->cmd_dma_state = true; + base->use_posbuf = 1; + base->bdl_pos_adj = 0; + base->sync_write = 1; + bus->pci = pdev; + bus->modelname = model; + bus->mixer_assigned = -1; + mutex_init(&bus->prepare_mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); +EXPORT_SYMBOL_GPL(snd_hda_ext_bus_init); /** * snd_hdac_ext_bus_exit - clean up a HD-audio extended bus diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5b1a15e39912..95de41d14e56 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -886,16 +886,9 @@ static int skl_create(struct pci_dev *pci, #if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) ext_ops = snd_soc_hdac_hda_get_ops(); #endif - snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); - bus->use_posbuf = 1; + snd_hda_ext_bus_init(hbus, pci, NULL, ext_ops, "sklbus"); skl->pci = pci; INIT_WORK(&skl->probe_work, skl_probe_work); - bus->bdl_pos_adj = 0; - - mutex_init(&hbus->prepare_mutex); - hbus->pci = pci; - hbus->mixer_assigned = -1; - hbus->modelname = "sklbus"; *rskl = skl; diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 30025d3c16b6..5d5081f80e88 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -8,6 +8,7 @@ // Authors: Keyon Jie <yang.jie(a)linux.intel.com> #include <linux/io.h> +#include <linux/pci.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include "../sof-priv.h" @@ -53,24 +54,25 @@ static const struct hdac_bus_ops bus_core_ops = { /* * This can be used for both with/without hda link support. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev) +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model) { #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - snd_hdac_ext_bus_init(bus, dev, &bus_core_ops, sof_hda_ext_ops); + snd_hda_ext_bus_init(bus, pdev, &bus_core_ops, sof_hda_ext_ops, model); #else /* CONFIG_SND_SOC_SOF_HDA */ memset(bus, 0, sizeof(*bus)); - bus->dev = dev; + bus->core.dev = &pdev->dev; - INIT_LIST_HEAD(&bus->stream_list); + INIT_LIST_HEAD(&bus->core.stream_list); - bus->irq = -1; + bus->core.irq = -1; /* * There is only one HDA bus atm. keep the index as 0. * Need to fix when there are more than one HDA bus. */ - bus->idx = 0; + bus->core.idx = 0; - spin_lock_init(&bus->reg_lock); + spin_lock_init(&bus->core.reg_lock); #endif /* CONFIG_SND_SOC_SOF_HDA */ } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fbc2421c77f8..03a68d286c7c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -609,8 +609,9 @@ static int hda_init(struct snd_sof_dev *sdev) bus = sof_to_bus(sdev); /* HDA bus init */ - sof_hda_bus_init(bus, &pci->dev); + sof_hda_bus_init(hbus, pci, hda_model); +#if !IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) bus->use_posbuf = 1; bus->bdl_pos_adj = 0; bus->sync_write = 1; @@ -619,7 +620,7 @@ static int hda_init(struct snd_sof_dev *sdev) hbus->pci = pci; hbus->mixer_assigned = -1; hbus->modelname = hda_model; - +#endif /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); bus->remap_addr = pci_ioremap_bar(pci, 0); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 1195018a1f4f..a4ec88f36512 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -635,7 +635,8 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev); /* * HDA bus operations. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev); +void sof_hda_bus_init(struct hda_bus *bus, struct pci_dev *pdev, + const char *model); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /* -- 2.25.1

3 11

[PATCH] Revert "ASoC: rockchip: add config for rockchip dmaengine pcm register"
by Sugar Zhang 21 Oct '21

21 Oct '21

This reverts commit 75b31192fe6ad20b42276b20ee3bdf1493216d63. The original purpose of customized pcm was to config prealloc buffer size flexibly. but, we can do the same thing by soc-generic-dmaengine-pcm. And the generic one can generated the better config by querying DMA capabilities from dmaengine driver rather than the Hard-Coded one. e.g. the customized one: static const struct snd_pcm_hardware snd_rockchip_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, ... the generic one: ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause && dma_caps.cmd_resume) hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw.info |= SNDRV_PCM_INFO_BATCH; ... So, let's revert back to use the generic dmaengine pcm. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/Makefile | 3 +-- sound/soc/rockchip/rockchip_i2s.c | 3 +-- sound/soc/rockchip/rockchip_pcm.c | 44 --------------------------------------- sound/soc/rockchip/rockchip_pcm.h | 11 ---------- 4 files changed, 2 insertions(+), 59 deletions(-) delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 65e814d..05b078e 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,11 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support snd-soc-rockchip-i2s-objs := rockchip_i2s.o -snd-soc-rockchip-pcm-objs := rockchip_pcm.o snd-soc-rockchip-pdm-objs := rockchip_pdm.o snd-soc-rockchip-spdif-objs := rockchip_spdif.o -obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o snd-soc-rockchip-pcm.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 7e89f5b..a6d7656 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -20,7 +20,6 @@ #include <sound/dmaengine_pcm.h> #include "rockchip_i2s.h" -#include "rockchip_pcm.h" #define DRV_NAME "rockchip-i2s" @@ -756,7 +755,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = rockchip_pcm_platform_register(&pdev->dev); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_suspend; diff --git a/sound/soc/rockchip/rockchip_pcm.c b/sound/soc/rockchip/rockchip_pcm.c deleted file mode 100644 index 02254e4..0000000 --- a/sound/soc/rockchip/rockchip_pcm.c +++ /dev/null @@ -1,44 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/dmaengine_pcm.h> - -#include "rockchip_pcm.h" - -static const struct snd_pcm_hardware snd_rockchip_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 1, - .periods_max = 52, - .buffer_bytes_max = 64 * 1024, - .fifo_size = 32, -}; - -static const struct snd_dmaengine_pcm_config rk_dmaengine_pcm_config = { - .pcm_hardware = &snd_rockchip_hardware, - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .prealloc_buffer_size = 32 * 1024, -}; - -int rockchip_pcm_platform_register(struct device *dev) -{ - return devm_snd_dmaengine_pcm_register(dev, &rk_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); -} -EXPORT_SYMBOL_GPL(rockchip_pcm_platform_register); - -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/rockchip/rockchip_pcm.h b/sound/soc/rockchip/rockchip_pcm.h deleted file mode 100644 index 7f00e2c..0000000 --- a/sound/soc/rockchip/rockchip_pcm.h +++ /dev/null @@ -1,11 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright (c) 2018 Rockchip Electronics Co. Ltd. - */ - -#ifndef _ROCKCHIP_PCM_H -#define _ROCKCHIP_PCM_H - -int rockchip_pcm_platform_register(struct device *dev); - -#endif -- 2.7.4

3 2

[PATCH 0/2] ASoC: qcom: sm8250: add support for TX and RX Macro dais
by Srinivas Kandagatla 21 Oct '21

21 Oct '21

This patchset adds support for testing WCD938X connected via TX and RX Macros on SM8250 MTP. Srinivas Kandagatla (2): ASoC: qcom: sm8250: add support for TX and RX Macro dais ASoC: qcom: sm8250: Add Jack support sound/soc/qcom/sm8250.c | 79 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 79 insertions(+) -- 2.21.0

2 3

a52 plugin not working with PipeWire
by GitHub issues - edited 21 Oct '21

21 Oct '21

alsa-project/alsa-plugins issue #37 was edited from sjnewbury: Using the lastest code since v1.2.5 I can successfully get AC3 output through HDMI when using speaker-test. Defaults use a huge buffer and gives terrible latency, but it works. PulseAudio also works fine with fairly low-latency using "tsched=1 tsched_buffer_size=32768 tsched_buffer_watermark=8192 fixed_latency_range=yes mmap=1". This doesn't need the rate kludge any more. When I try pipewire using mmap I get the below messages in the log, no audio: `spa.alsa: alsa-pcm a52hdmi: mmap_commit wrote 512 instead of 1024` With mmap disabled, no errors are reported, but still no sound. Is there a configuration for pipewire which works with the a52 plugin, a pipewire bug, or is this another a52 plugin bug? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/37 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

a52 plugin not working with PipeWire
by GitHub issues - opened 21 Oct '21

21 Oct '21

alsa-project/alsa-plugins issue #37 was opened from sjnewbury: Using the lastest code since v1.2.5 I can successfully get AC3 output through HDMI when using speaker-test. Defaults uses a huge buffer and gives terrible latency, but it works. PulseAudio also works fine with fairly low-latency using "tsched=1 tsched_buffer_size=32768 tsched_buffer_watermark=8192 fixed_latency_range=yes mmap=1". This doesn't need the rate kludge any more. When I try pipewire using mmap I get the below messages in the log, no audio: `spa.alsa: alsa-pcm a52hdmi: mmap_commit wrote 512 instead of 1024` With mmap disabled, no errors are reported, but still no sound. Is there a configuration for pipewire which works with the a52 plugin, a pipewire bug, or is this another a52 plugin bug? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/37 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek: Fix mic mute LED for the HP Spectre x360 14
by Johnathon Clark 21 Oct '21

21 Oct '21

On the 'HP Spectre x360 Convertible 14-ea0xx' the microphone mute led is controlled by GPIO 0x04. The speaker mute LED does not seem to be exposed by GPIO and is there not set. Signed-off-by: Johnathon Clark <john.clark(a)cantab.net> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 965b096f416f..79c6fb39864d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4355,6 +4355,17 @@ static void alc287_fixup_hp_gpio_led(struct hda_codec *codec, alc_fixup_hp_gpio_led(codec, action, 0x10, 0); } +static void alc245_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->micmute_led_polarity = 1; + } + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + /* turn on/off mic-mute LED per capture hook via VREF change */ static int vref_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -6709,6 +6720,7 @@ enum { ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, ALC287_FIXUP_HP_GPIO_LED, ALC256_FIXUP_HP_HEADSET_MIC, + ALC245_FIXUP_HP_GPIO_LED, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, @@ -7333,6 +7345,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC245_FIXUP_HP_X360_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_x360_amp, + .chained = true, + .chain_id = ALC245_FIXUP_HP_GPIO_LED }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, @@ -8432,6 +8446,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_tongfang_reset_persistent_settings, }, + [ALC245_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 2.33.1

2 1

[PATCH v2 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 21 Oct '21

21 Oct '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 139 +++++++++++++++ sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 164 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 29 ++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 23 ++- 5 files changed, 354 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

1 2

[PATCH AUTOSEL 4.4 6/7] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 4727f5b80e76..4ee3458ad810 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -288,8 +288,9 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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[PATCH AUTOSEL 4.9 6/7] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 00c6af2ae1c2..f0e112906c68 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -389,8 +389,9 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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[PATCH AUTOSEL 4.14 8/9] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 778b42ba90b8..5ae72134159a 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -389,8 +389,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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[PATCH AUTOSEL 4.19 09/10] ALSA: hda: avoid write to STATESTS if controller is in reset
by Sasha Levin 21 Oct '21

21 Oct '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit b37a15188eae9d4c49c5bb035e0c8d4058e4d9b3 ] The snd_hdac_bus_reset_link() contains logic to clear STATESTS register before performing controller reset. This code dates back to an old bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio codec probing robustness"). Originally the code was added to azx_reset(). The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus into controller object") and ended up to snd_hdac_bus_reset_link() and called primarily via snd_hdac_bus_init_chip(). The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is called when controller is not in reset. In this case, STATESTS can be cleared. This can be useful e.g. when forcing a controller reset to retry codec probe. A normal non-power-on reset will not clear the bits. However, this old logic is problematic when controller is already in reset. The HDA specification states that controller must be taken out of reset before writing to registers other than GCTL.CRST (1.0a spec, 3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost if the controller is already in reset per the HDA specification mentioned. This has been harmless on older hardware. On newer generation of Intel PCIe based HDA controllers, if configured to report issues, this write will emit an unsupported request error. If ACPI Platform Error Interface (APEI) is enabled in kernel, this will end up to kernel log. Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if the function is called when controller is not in reset. Otherwise clearing the bits is not possible and should be skipped. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_controller.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 74244d8e2909..a65e8c0c630d 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -390,8 +390,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); -- 2.33.0

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