July 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Incorrect name of PCM
by Christopher Head 24 Jul '18

24 Jul '18

Hi folks, I have an Asus P8H77-I motherboard. On board is an Intel HDA connected to a Via VT1708S codec. There are a number of analogue ports. In addition to them, on the back panel are an HDMI port (from an integrated GPU) and a TOSLINK port. The motherboard manual isn’t totally clear on whether the HDMI port can also carry sound or only video, but Windows claims to be able to send sound there, so probably yes. I don’t use the integrated GPU, though. Here’s the problem: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1708S Alt Analog [VT1708S Alt Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: VT1708S Digital [VT1708S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 OK, a little bit unexpected that there is only one digital device listed rather than two. The codec datasheet says it should have two digital output widgets. Maybe it’s due to the integrated GPU being disabled though. $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Digital HDMI Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 3 HDMI Audio Output Why is this wrong? Because when I play to the “hdmi” PCM, at least with the S/PDIF switch turned on in mixer controls, the sound comes out the TOSLINK port! It seems that the PCM naming is wrong. This is all without any configuration file. Everything works fine, just the name is wrong. I don’t know whether turning off the S/PDIF switch results in the sound coming out the HDMI port instead; maybe it does (I don’t have anything that speaks HDMI and can also play sound to test with). Either way, seems like it should be called either “spdif” (if it’s only the TOSLINK port) or some combination of the two (if it’s both and switchable by the mixer switch). If replying, please note that I’m not subscribed. Thanks! -- Christopher Head

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Re: [alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Mark Brown 24 Jul '18

24 Jul '18

On Tue, Jul 24, 2018 at 01:49:48AM +0000, Bard Liao wrote: > Please ignore these patches. There is no significant advantage of > adding "Volume" to the control name. To prevent breaking the > configuration files on user space, we should find out all projects > which are using these controls and modify them. It will take time. > So I think it is better to leave it as it is. The RT5631 doesn't appear to have any users except Arndale, certainly no x86 users?

1 0

[alsa-devel] Applied "ASoC: doc: replace codec to component" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: doc: replace codec to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a3ac7727f3a226b0569431de62a4e9bb4c770b2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:14:22 +0000 Subject: [PATCH] ASoC: doc: replace codec to component Now we can replace Codec to Component. Let's do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index f87612b94812..58f625fe3d39 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -179,12 +179,12 @@ i.e. static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; if (mute) - snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else - snd_soc_write(codec, WM8974_DAC, mute_reg); + snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: trace: remove snd_soc_codec" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: trace: remove snd_soc_codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 211b4415ecb254e167837206756dea123342689c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:18:13 +0000 Subject: [PATCH] ASoC: trace: remove snd_soc_codec snd_soc_codec is replaced to snd_soc_component, and it is not used in this file. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/trace/events/asoc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ccd1a3bdff46..40c300fe704d 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -12,7 +12,6 @@ #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") struct snd_soc_jack; -struct snd_soc_codec; struct snd_soc_card; struct snd_soc_dapm_widget; struct snd_soc_dapm_path; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: meson: axg-spdifout: select SND_PCM_IEC958
by Arnd Bergmann 24 Jul '18

24 Jul '18

When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the new driver: sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params': axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params' The other users use 'select', so we should do the same here. Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 4cf93c05a982..8af8bc358a90 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -56,6 +56,7 @@ config SND_MESON_AXG_SOUND_CARD config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 imply SND_SOC_SPDIF help Select Y or M to add support for SPDIF output serializer embedded -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied
by Oder Chiou 24 Jul '18

24 Jul '18

The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91..dca82dd 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

2 1

[alsa-devel] WARNING in port_delete
by Dae R. Jeong 24 Jul '18

24 Jul '18

Reporting the crash: WARNING in port_delete This crash has been found in v4.18-rc3 using RaceFuzzer (a modified version of Syzkaller), which we descrbie more at the end of this report. Our analysis shows that the race occurs when invoking two close syscalls concurrently. The executed program is as follows: r0 = open("/dev/snd/seq", 0x0, 0x0) ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT, {{0x80}, "706f72943000000000000000000000000000000000000000000000000000000000233f770800000000000000000000000000000000000000001000", 0xffffffffffffffff, 0xfffffffffffffbff}) r1 = openat(AT_FDCWD, "/dev/sequencer2", 0x8402, 0x0) close(r0) close(r1) and two threads executed the program as follows: Thread0 Thread1 open("/dev/snd/seq") ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT) openat("/dev/sequencer2") close(r0) close(r1) (Note that two close() syscalls were exeuted after openat() syscall, and two close() syscalls were executed concurrently.) Crash log: ================================================================== WARNING: CPU: 1 PID: 32519 at /home/daeryong/workspace/race-fuzzer/kernels_repo/kernel_v4.18-rc3/sound/core/seq/seq_ports.c:275 port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 32519 Comm: syz-executor0 Not tainted 4.18.0-rc3 #1 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x16e/0x22c lib/dump_stack.c:113 panic+0x1a8/0x3a7 kernel/panic.c:184 __warn+0x191/0x1a0 kernel/panic.c:536 report_bug+0x132/0x1b0 lib/bug.c:186 fixup_bug.part.10+0x28/0x50 arch/x86/kernel/traps.c:178 fixup_bug arch/x86/kernel/traps.c:247 [inline] do_error_trap+0x284/0x2c0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Code: 00 00 e8 b5 68 d2 fd 8b 83 58 01 00 00 85 c0 75 1d e8 86 5c ae fd 48 89 df e8 3e 52 d2 fd 31 c0 5b 41 5c 5d c3 e8 72 5c ae fd <0f> 0b eb c8 e8 69 5c ae fd 0f 0b eb da 0f 1f 44 00 00 55 48 89 e5 RSP: 0018:ffff8801d635f900 EFLAGS: 00010216 RAX: 0000000000040000 RBX: ffff8801da196900 RCX: ffffffff839b86ee RDX: 0000000000005e94 RSI: ffffc90002640000 RDI: ffff8801da196978 RBP: ffff8801d635f910 R08: 0000000000000098 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffffffff839ba820 R13: ffff8801da196950 R14: ffff8801ed5bc080 R15: ffff8801da196958 snd_seq_delete_port+0x2b0/0x2d0 sound/core/seq/seq_ports.c:303 snd_seq_ioctl_delete_port+0x5b/0xa0 sound/core/seq/seq_clientmgr.c:1325 snd_seq_kernel_client_ctl+0xd4/0xf0 sound/core/seq/seq_clientmgr.c:2361 snd_seq_event_port_detach+0xcb/0x120 sound/core/seq/seq_ports.c:705 delete_port+0x3a/0x70 sound/core/seq/oss/seq_oss_init.c:354 snd_seq_oss_release+0x7c/0x90 sound/core/seq/oss/seq_oss_init.c:433 odev_release+0x40/0x60 sound/core/seq/oss/seq_oss.c:153 __fput+0x234/0x470 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x15a/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:192 [inline] exit_to_usermode_loop+0x2a3/0x2b0 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x485/0x4b0 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x456469 Code: 1d ba fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 eb b9 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007fe9a8936b28 EFLAGS: 00000246 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 000000000072bfa0 RCX: 0000000000456469 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000014 RBP: 0000000000000054 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007fe9a89376d4 R13: 00000000ffffffff R14: 00000000006f5880 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. = About RaceFuzzer RaceFuzzer is a customized version of Syzkaller, specifically tailored to find race condition bugs in the Linux kernel. While we leverage many different technique, the notable feature of RaceFuzzer is in leveraging a custom hypervisor (QEMU/KVM) to interleave the scheduling. In particular, we modified the hypervisor to intentionally stall a per-core execution, which is similar to supporting per-core breakpoint functionality. This allows RaceFuzzer to force the kernel to deterministically trigger racy condition (which may rarely happen in practice due to randomness in scheduling). RaceFuzzer's C repro always pinpoints two racy syscalls. Since C repro's scheduling synchronization should be performed at the user space, its reproducibility is limited (reproduction may take from 1 second to 10 minutes (or even more), depending on a bug). This is because, while RaceFuzzer precisely interleaves the scheduling at the kernel's instruction level when finding this bug, C repro cannot fully utilize such a feature. Please disregard all code related to "should_hypercall" in the C repro, as this is only for our debugging purposes using our own hypervisor.

3 4

[alsa-devel] [PATCH 0/4] drm_audio_component support for AMD/ATI HDMI codecs
by Takashi Iwai 24 Jul '18

24 Jul '18

Hi, this is a patch set to add the support of drm_audio_component for AMD/ATI HDMI codecs. With the drm_audio_component, the HDMI/DP audio hotplug and ELD read-out can be achieved directly without the hardware access. The best point by that is that it makes the hotplug notification working even during runtime suspend. The support is totally optional and dynamic, hence it still works even if either HD-audio or DRM side isn't patched, and it'll fall back to the existing method. The current patch supports only radeon and a part of amdgpu; the DC support isn't included yet. Takashi === Takashi Iwai (4): ALSA: hda/hdmi: Use single mutex unlock in error paths ALSA: hda/hdmi: Allow audio component for AMD/ATI HDMI drm/radeon: Add audio component support drm/amdgpu: Add audio component support drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/amdgpu/Makefile | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 4 + drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c | 97 ++++++++++ drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h | 3 + drivers/gpu/drm/amd/amdgpu/dce_v10_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v11_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v6_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v8_0.c | 6 + drivers/gpu/drm/radeon/radeon.h | 3 + drivers/gpu/drm/radeon/radeon_audio.c | 79 ++++++++ sound/pci/hda/patch_hdmi.c | 209 +++++++++++++++++----- 12 files changed, 374 insertions(+), 49 deletions(-) create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c -- 2.18.0

2 6

[alsa-devel] Applied "ASoC: rockchip-i2s: add description for px30" to the asoc tree
by Mark Brown 23 Jul '18

23 Jul '18

The patch ASoC: rockchip-i2s: add description for px30 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2f562a4739602d96928de0b040ce46b00d8d3adc Mon Sep 17 00:00:00 2001 From: Liang Chen <cl(a)rock-chips.com> Date: Mon, 23 Jul 2018 17:25:21 +0800 Subject: [PATCH] ASoC: rockchip-i2s: add description for px30 Add "rockchip,px30-i2s", "rockchip,rk3066-i2s" for i2s on px30 platform. Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Liang Chen <cl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b208a752576c..54aefab71f2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,6 +7,7 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 -- 2.18.0

1 0

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..06ae3f5adf4b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1443,6 +1443,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.18.0

2 1

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

[alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Yue Wang 23 Jul '18

23 Jul '18

This patch adds the support for native DSD for Topping DX7s DAC. The product description is on http://en.tpdz.net/products_detail/productId=32.html and it's available on https://www.amazon.com/Balanced-Headphone-Amplifier-Topping-2ES9038Q2M/dp/B… The DAC uses XMOS chip but uses a special vendor id and product id. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

Re: [alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Takashi Iwai 23 Jul '18

23 Jul '18

On Mon, 23 Jul 2018 09:25:34 +0200, <yuleopen(a)gmail.com> wrote: > > From: Yue Wang <yuleopen(a)gmail.com> > > Signed-off-by: Yue Wang <yuleopen(a)gmail.com> Please give more explanation. No commit message already indicates that something is very wrong, no matter what the change is. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: core: fix unsigned int pages overflow when comapred
by He, Bo 23 Jul '18

23 Jul '18

we see the below kernel panic on stress suspend resume test in snd_malloc_sgbuf_pages(), snd_dma_alloc_pages_fallback() alloc chunk maybe larger than the left pages due to the pages alignment, which will cause the pages overflow. while (pages > 0) { ... pages -= chunk; } the patch is change the pages from unsigned int to int to fix the issue. BUG: unable to handle kernel paging request at ffff88000deb4000 IP: [<ffffffff81404fa9>] memset_erms+0x9/0x10 Call Trace: [<ffffffff818f222f>] snd_dma_alloc_pages+0xff/0x210 [<ffffffff818f23af>] snd_dma_alloc_pages_fallback+0x6f/0x90 [<ffffffff818f2b85>] snd_malloc_sgbuf_pages+0x145/0x370 [<ffffffff818f229e>] snd_dma_alloc_pages+0x16e/0x210 [<ffffffffc011930d>] hdac_ext_dma_alloc_pages+0x1d/0x40 [snd_hda_ext_core] [<ffffffffc010729a>] snd_hdac_dsp_prepare+0xca/0x1c0 [snd_hda_core] [<ffffffffc01880f9>] skl_dsp_prepare+0x99/0xf0 [snd_soc_skl] [<ffffffffc0162a7e>] bxt_load_base_firmware+0x9e/0x5c0 [snd_soc_skl_ipc] [<ffffffffc01630ec>] bxt_set_dsp_D0+0x14c/0x300 [snd_soc_skl_ipc] [<ffffffffc015f9c3>] skl_dsp_get_core+0x43/0xd0 [snd_soc_skl_ipc] [<ffffffffc015fa60>] skl_dsp_wake+0x10/0x20 [snd_soc_skl_ipc] [<ffffffffc0188e3e>] skl_resume_dsp+0x7e/0x140 [snd_soc_skl] [<ffffffffc0183c4a>] skl_resume+0xda/0x170 [snd_soc_skl] [<ffffffff81452726>] pci_pm_resume+0x76/0xe0 [<ffffffff816616da>] dpm_run_callback+0x5a/0x180 [<ffffffff81661e3c>] device_resume+0xdc/0x2c0 [<ffffffff81663818>] dpm_resume+0x118/0x310 [<ffffffff81663e11>] dpm_resume_end+0x11/0x20 [<ffffffff810f8bcc>] suspend_devices_and_enter+0x11c/0x2b0 [<ffffffff810f90bd>] pm_suspend+0x35d/0x3d0 [<ffffffff810f78a6>] state_store+0x66/0x90 [<ffffffff813f80e2>] kobj_attr_store+0x12/0x20 [<ffffffff812a37bc>] sysfs_kf_write+0x3c/0x50 [<ffffffff812a2cbd>] kernfs_fop_write+0x11d/0x1a0 [<ffffffff8121dfaa>] __vfs_write+0x3a/0x150 [<ffffffff8121f2b1>] vfs_write+0xb1/0x1a0 [<ffffffff81220898>] SyS_write+0x58/0xc0 [<ffffffff81001fca>] do_syscall_64+0x6a/0xe0 [<ffffffff81b06560>] entry_SYSCALL_64_after_swapgs+0x5d/0xd7 Signed-off-by: he, bo <bo.he(a)intel.com> Signed-off-by: zhang jun <jun.zhang(a)intel.com> --- sound/core/sgbuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffab..33449ee 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -68,7 +68,8 @@ void *snd_malloc_sgbuf_pages(struct device *device, size_t *res_size) { struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; + unsigned int i, chunk, maxpages; + int pages; struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; -- 2.7.4

3 7

[alsa-devel] [PATCH v6 00/10] soundwire: Add multi link support
by Sanyog Kale 23 Jul '18

23 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch - Fix to avoid bus lock acquired twice changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes changes in v5: - Add patch to fix duplicate stream state assignment - fix incorrect exit which was releasing the stream after configuring master - In sdw_release_bus_lock(), release the mutex in reverse order as suggested by Takashi changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters Sanyog Kale (3): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances soundwire: Fix acquiring bus lock twice during master release Shreyas NC (6): soundwire: Initialize completion for defer messages soundwire: Fix duplicate stream state assignment soundwire: fix incorrect exit after configuring stream soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.13.0

3 14

[alsa-devel] [PATCH] ASoC: wm9712: fix replace codec to component
by Marcel Ziswiler 23 Jul '18

23 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Since commit 143b44845d87 ("ASoC: wm9712: replace codec to component") "wm9712-codec" got renamed to "wm9712-component", however, this change never got propagated down to the actual board/platform drivers. E.g. on Colibri T20 this lead to the following spew upon boot with sound/touch being broken: [ 2.214121] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.222137] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.344384] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.351885] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.668339] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.675811] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.208408] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.216312] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.235397] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.248938] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 14.970443] ALSA device list: [ 14.996628] No soundcards found. This commit finally fixes this again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- arch/arm/mach-pxa/tosa.c | 2 +- arch/mips/alchemy/devboards/db1300.c | 2 +- drivers/mfd/wm97xx-core.c | 6 +++--- sound/soc/au1x/db1200.c | 4 ++-- sound/soc/fsl/pcm030-audio-fabric.c | 6 +++--- sound/soc/fsl/phycore-ac97.c | 6 +++--- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 4 ++-- 11 files changed, 23 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-pxa/tosa.c b/arch/arm/mach-pxa/tosa.c index cb5cd8e78c94..32011e600496 100644 --- a/arch/arm/mach-pxa/tosa.c +++ b/arch/arm/mach-pxa/tosa.c @@ -848,7 +848,7 @@ static struct platform_device sharpsl_rom_device = { }; static struct platform_device wm9712_device = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = -1, }; diff --git a/arch/mips/alchemy/devboards/db1300.c b/arch/mips/alchemy/devboards/db1300.c index efb318e03e0a..841a27736eb1 100644 --- a/arch/mips/alchemy/devboards/db1300.c +++ b/arch/mips/alchemy/devboards/db1300.c @@ -638,7 +638,7 @@ static struct platform_device db1300_sd0_dev = { /**********************************************************************/ static struct platform_device db1300_wm9715_dev = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = 1, /* ID of PSC for AC97 audio, see asoc glue! */ }; diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index f5a8347f837f..9fec11fc0410 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -109,7 +109,7 @@ static const struct regmap_config wm9705_regmap_config = { }; static struct mfd_cell wm9705_cells[] = { - { .name = "wm9705-codec", }, + { .name = "wm9705-component", }, { .name = "wm97xx-ts", }, }; @@ -177,7 +177,7 @@ static const struct regmap_config wm9712_regmap_config = { }; static struct mfd_cell wm9712_cells[] = { - { .name = "wm9712-codec", }, + { .name = "wm9712-component", }, { .name = "wm97xx-ts", }, }; @@ -248,7 +248,7 @@ static const struct regmap_config wm9713_regmap_config = { }; static struct mfd_cell wm9713_cells[] = { - { .name = "wm9713-codec", }, + { .name = "wm9713-component", }, { .name = "wm97xx-ts", }, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 301e1fc9a377..d452478d62b4 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .codec_dai_name = "ac97-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "ac97-codec.1", + .codec_name = "ac97-component.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -68,7 +68,7 @@ static struct snd_soc_dai_link db1300_ac97_dai = { .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "wm9712-codec.1", + .codec_name = "wm9712-component.1", }; static struct snd_soc_card db1300_ac97_machine = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..869f4b226ccd 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -33,14 +33,14 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; @@ -85,7 +85,7 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "request_module returned: %d\n", ret); - pdata->codec_device = platform_device_alloc("wm9712-codec", -1); + pdata->codec_device = platform_device_alloc("wm9712-component", -1); if (!pdata->codec_device) dev_err(&op->dev, "platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..dbd81d3b7c2c 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -30,8 +30,8 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { { .name = "HiFi", .stream_name = "HiFi", - .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-component", .cpu_dai_name = "imx-ssi.0", .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, @@ -88,7 +88,7 @@ static int __init imx_phycore_init(void) if (ret) goto fail1; - imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); + imx_phycore_snd_device = platform_device_alloc("wm9712-component", -1); if (!imx_phycore_snd_device) { ret = -ENOMEM; goto fail2; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..937746d1e23e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -83,7 +83,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..e958c4c241f6 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..040db695270e 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -93,7 +93,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, @@ -102,7 +102,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..af42417ccec5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, { @@ -198,7 +198,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, }; diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a3345972e..aa12be76d15a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -53,7 +53,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .init = tegra_wm9712_init, }; @@ -83,7 +83,7 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, machine); - machine->codec = platform_device_alloc("wm9712-codec", -1); + machine->codec = platform_device_alloc("wm9712-component", -1); if (!machine->codec) { dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); return -ENOMEM; -- 2.14.4

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[alsa-devel] [PATCH] ALSA: hda/i915: Allow delayed i915 audio component binding
by Takashi Iwai 22 Jul '18

22 Jul '18

Currently HD-audio i915 audio binding doesn't support any delayed binding, and supposes that the i915 driver registers the component immediately. This has been OK, so far, but the work-in-progress change in i915 may introduce the asynchronous binding, which effectively delays the component registration. For addressing it, implement a completion to be synced with the master binding. The timeout is set to 10 seconds which should be long enough and hopefully be not too annoying if anyone boots up a debugging session with i915 KMS turned off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is applied on top of the current for-next or topic/drm_audio_component branch of sound git tree. sound/hda/hdac_i915.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8f2aa8bc1185..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,8 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> +static struct completion bind_complete; + #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -97,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -117,7 +132,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - err = snd_hdac_acomp_init(bus, NULL, + init_completion(&bind_complete); + + err = snd_hdac_acomp_init(bus, &i915_init_ops, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -125,8 +142,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) acomp = bus->audio_component; if (!acomp) return -ENODEV; - if (!acomp->ops) + if (!acomp->ops) { request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } if (!acomp->ops) { snd_hdac_acomp_exit(bus); return -ENODEV; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 21 Jul '18

21 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- This patch tries to solve the same problem as in the following patches, but does so in a way that doesn't require registering fixed regulators in the audio machine-driver: https://patchwork.kernel.org/patch/10531097 & 10531099 https://patchwork.kernel.org/patch/10536023/ include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..d8f202c57f958d 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (!device_property_read_string(dev, "dlg,vddio", &of_val32)) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

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[alsa-devel] [asoc:for-4.19 199/204] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 21 Jul '18

21 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v1 0/7] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 20 Jul '18

20 Jul '18

From: Liang Chen <cl(a)rock-chips.com> Liang Chen (7): Documentation: usb: dwc2: add description for px30 Documentation: rockchip-dw-mshc: add description for px30 Documentation: spi-rockchip: add description for px30 Documentation: rockchip-i2s: add description for px30 Documentation: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/mmc/rockchip-dw-mshc.txt | 1 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/spi/spi-rockchip.txt | 1 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 539 ++++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2044 ++++++++++++++++++++ 9 files changed, 2594 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: pxa: remove clock divider and pll setup from zylonite and magician
by Daniel Mack 20 Jul '18

20 Jul '18

The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/magician.c | 106 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 2 insertions(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: topology: Increase SND_SOC_TPLG_NUM_TEXTS to 32
by Puneeth Prabhu 20 Jul '18

20 Jul '18

Topology supports maximum of 32 configs for a path whereas kernel can support only 16 configs of a path. So, increase SND_SOC_TPLG_NUM_TEXTS to 32 to support 32 configs for a path. Signed-off-by: Puneeth Prabhu <puneethx.prabhu(a)intel.com> Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index a74ca23..5b1411b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -88,7 +88,7 @@ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ /* string sizes */ -#define SND_SOC_TPLG_NUM_TEXTS 16 +#define SND_SOC_TPLG_NUM_TEXTS 32 /* ABI version */ #define SND_SOC_TPLG_ABI_VERSION 0x5 /* current version */ -- 2.1.4

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[alsa-devel] No controls in alsamixer
by Riccardo Casagrande 20 Jul '18

20 Jul '18

Hello all, following this thread http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/108068.html I added a new sound card without codec. On my board the i.MX6Q chip puts audio data directly to some ST FDA2100 power amplifiers. The sound card is listed correctly in alsa but in alsamixer there are no audio controls, I need a volume control. I tried to edit asound.conf file with this part pcm.softvol { type softvol slave { pcm "hw:1,0" } control { name "SoftVol" card 1 } } ctl.softvol { type hw card 1 } This is part of logs during the boot --> ITEL imx_itel_probe --> ITEL Configuro audmux --> ITEL imx-audmux-confic ext-port port=0x00000001ptcr=0x9ce73800 pdcr=0x00006000 --> ITEL imx-audmux-config int-port port=0x00000001ptcr=0x00000800 pdcr=0x00002000 --> ITEL imx-probe registro card imx-itel sound-audio-itel: snd-soc-dummy-dai <-> 202c000.ssi mapping ok --> ITEL imx-probe of ssi --> ITEL imx-probe of codec [...] ALSA device list: #0: cs42888-audio #1: imx-audio-itel #2: imx-spdif #3: imx-hdmi-soc [...] ALSA: Restoring mixer settings... No state is present for card imxaudioitel Found hardware: "imx-audio-itel" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxaudioitel No state is present for card imxspdif Found hardware: "imx-spdif" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxspdif Sorry but I'm new in alsa and linux in general. Riccardo

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[alsa-devel] [PATCH] ASoC: sgtl5000: fix spelling in devicetree binding document
by Marcel Ziswiler 20 Jul '18

20 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> This fixes a spelling mistake. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0f214457476f..9c58f724396a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -17,7 +17,7 @@ Optional properties: - VDDD-supply : the regulator provider of VDDD -- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then -- 2.14.4

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[alsa-devel] [PATCH] MAINTAINERS: add entry for STI audio drivers
by Arnaud Pouliquen 20 Jul '18

20 Jul '18

Add sound/soc/sti drivers entry for STI audio drivers from ST Microelectronics. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 192d7f7..50a7a53 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13574,6 +13574,13 @@ L: linux-block(a)vger.kernel.org S: Maintained F: drivers/block/skd*[ch] +STI AUDIO (ASoC) DRIVERS +M: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +F: sound/soc/sti/ + STI CEC DRIVER M: Benjamin Gaignard <benjamin.gaignard(a)linaro.org> S: Maintained -- 2.7.4

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