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

Patch For Lenovo Legion 7 15IMHg05 & Yoga 7 14ITL5
by Cameron Berkenpas 08 Jul '21

08 Jul '21

Hello, I have a patch against 5.12.x that results in working speaker audio for the Lenovo Legion 7 15IMHg05 and the Lenovo Yoga 7 14ITL5. How should I go about getting this merged? Should this be against the latest torvalds git? Is this the right place to present this particular patch? There are no issues checking my changes against checkpatch.pl. Thanks! diff --git a/sound/pci/hda/legion_15imhg05_helper.c b/sound/pci/hda/legion_15imhg05_helper.c new file mode 100644 index 000000000000..6ead999ec3c9 --- /dev/null +++ b/sound/pci/hda/legion_15imhg05_helper.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Legion 15IMHg05 speaker output. */ + +static const struct hda_verb legion_15imhg05_coefs[] = { +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x41 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, + +// Set right speaker for Legion 7i. +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0xc }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2a }, +{0x20, 0x4b0, 0x20 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, +{} +}; + +static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_INIT: + spec->gen.automute_speaker = 0; + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e46e43dac6bf..28a4955cf2c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6352,6 +6352,12 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" +/* For alc287_fixup_legion_15imhg05_speakers() */ +#include "legion_15imhg05_helper.c" + +/* For yoga7_14itl5_coefs */ +#include "yoga7_14itl5_coefs.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6561,6 +6567,9 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, + ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -8088,6 +8097,24 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = legion_15imhg05_coefs, + .chained = true, + .chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, + }, + [ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_legion_15imhg05_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, + [ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = yoga7_14itl5_coefs, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC295_FIXUP_ASUS_DACS] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_asus_dacs, @@ -8520,6 +8547,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), + SND_PCI_QUIRK(0x17aa, 0x3813, "Lenovo Legion 7", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), diff --git a/sound/pci/hda/yoga7_14itl5_coefs.c b/sound/pci/hda/yoga7_14itl5_coefs.c new file mode 100644 index 000000000000..a437fe042a1d --- /dev/null +++ b/sound/pci/hda/yoga7_14itl5_coefs.c @@ -0,0 +1,29 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Yoga 7 14ITL5 speaker output. */ + +static const struct hda_verb yoga7_14itl5_coefs[] = { +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x41 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, + +// set right speaker Yoga 7i. +{0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0xc }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2a }, +{0x20, 0x4b0, 0x20 }, +{0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x2 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, AC_VERB_SET_PROC_COEF, 0x0 }, +{0x20, 0x4b0, 0x20 }, +{} +}; +

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Audio out on DisplayPort but not HDMI, on AMD Ryzen APU/Vega
by Damjan Georgievski 08 Jul '21

08 Jul '21

I have a desktop computer with an AMD Ryzen 5 4650G PRO APU (CPU + on-die GPU). The motherboard (MSI Mortar B550m wifi / bios 1.6). has one displayport and one HDMI outputs. The displayport is connected to my Lenovo P27 monitor, and the HDMI to my Samsung TV. When I choose the GPU as an audio out, I only get audio on the 3.5" headphones jack of the Lenovo monitor, while I would like to get the audio on the Samsung TV over HDMI. I can't find a way to switch the different outputs (neither over pipewire or just using native alsa). pavucontrol configuration https://i.imgur.com/9nbVtNz.png alsamixer -c0 only has only one option to mute s/pdif, which mutes the output on the displayport -> Lenovo -> 3.5" jack. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Generic_1 [HD-Audio Generic], device 0: ALCS1200A Analog [ALCS1200A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Generic_1 [HD-Audio Generic], device 1: ALCS1200A Digital [ALCS1200A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Am I correct that card 1 here ^ should have at least two devices? distro is Archlinux kernel is 5.12.13-arch1-2 alsa-* 1.2.5.1-3 pipewire 1:0.3.30-2 plasma 5.22.2.1-1 xorg-server 1.20.11-1 amdgpu vanilla driver $ aplay -L null Discard all samples (playback) or generate zero samples (capture) sysdefault Default Audio Device samplerate Rate Converter Plugin Using Samplerate Library speexrate Rate Converter Plugin Using Speex Resampler jack JACK Audio Connection Kit oss Open Sound System pulse PulseAudio Sound Server speex Plugin using Speex DSP (resample, agc, denoise, echo, dereverb) upmix Plugin for channel upmix (4,6,8) vdownmix Plugin for channel downmix (stereo) with a simple spacialization default Default ALSA Output (currently PulseAudio Sound Server) usbstream:CARD=C93 罗技高清网络摄像机 C93 USB Stream Output hdmi:CARD=Generic,DEV=0 HD-Audio Generic, HDMI 0 HDMI Audio Output usbstream:CARD=Generic HD-Audio Generic USB Stream Output sysdefault:CARD=Generic_1 HD-Audio Generic, ALCS1200A Analog Default Audio Device front:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog Front output / input surround21:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Generic_1,DEV=0 HD-Audio Generic, ALCS1200A Digital IEC958 (S/PDIF) Digital Audio Output usbstream:CARD=Generic_1 HD-Audio Generic USB Stream Output $ lspci 00:00.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Root Complex 00:00.2 IOMMU: Advanced Micro Devices, Inc. [AMD] Renoir IOMMU 00:01.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:02.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:02.1 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe GPP Bridge 00:02.2 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe GPP Bridge 00:08.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir PCIe Dummy Host Bridge 00:08.1 PCI bridge: Advanced Micro Devices, Inc. [AMD] Renoir Internal PCIe GPP Bridge to Bus 00:14.0 SMBus: Advanced Micro Devices, Inc. [AMD] FCH SMBus Controller (rev 51) 00:14.3 ISA bridge: Advanced Micro Devices, Inc. [AMD] FCH LPC Bridge (rev 51) 00:18.0 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 0 00:18.1 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 1 00:18.2 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 2 00:18.3 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 3 00:18.4 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 4 00:18.5 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 5 00:18.6 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 6 00:18.7 Host bridge: Advanced Micro Devices, Inc. [AMD] Renoir Device 24: Function 7 16:00.0 USB controller: Advanced Micro Devices, Inc. [AMD] Device 43ee 16:00.1 SATA controller: Advanced Micro Devices, Inc. [AMD] Device 43eb 16:00.2 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43e9 20:08.0 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43ea 20:09.0 PCI bridge: Advanced Micro Devices, Inc. [AMD] Device 43ea 29:00.0 Network controller: Intel Corporation Wi-Fi 6 AX200 (rev 1a) 2a:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8125 2.5GbE Controller (rev 04) 2b:00.0 Non-Volatile memory controller: ADATA Technology Co., Ltd. XPG SX8200 Pro PCIe Gen3x4 M.2 2280 Solid State Drive (rev 03) 30:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] Renoir (rev d9) 30:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 30:00.2 Encryption controller: Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) Platform Security Processor 30:00.3 USB controller: Advanced Micro Devices, Inc. [AMD] Renoir USB 3.1 30:00.4 USB controller: Advanced Micro Devices, Inc. [AMD] Renoir USB 3.1 30:00.6 Audio device: Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller $ cat /proc/asound/devices 1: : sequencer 2: [ 1- 3]: digital audio playback 3: [ 1- 0]: hardware dependent 4: [ 1] : control 5: [ 0- 0]: digital audio capture 6: [ 0] : control 7: [ 2- 0]: digital audio playback 8: [ 2- 0]: digital audio capture 9: [ 2- 1]: digital audio playback 10: [ 2- 2]: digital audio capture 11: [ 2- 0]: hardware dependent 12: [ 2] : control 33: : timer $ cat /proc/asound/cards 0 [C93 ]: USB-Audio - 罗技高清网络摄像机 C93 罗技高清网络摄像机 C93 at usb-0000:30:00.3-1, high speed 1 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfcb88000 irq 95 2 [Generic_1 ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xfcb80000 irq 96 $ cat /proc/asound/card card0/ card1/ card2/ cards -- damjan

2 3

ALSA: intel8x0: div by zero in snd_intel8x0_update()
by Sergey Senozhatsky 08 Jul '21

08 Jul '21

Hi, I'm running (sometimes) into the following problem during resume divide error: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:snd_intel8x0_interrupt+0x121/0x279 Code: 42 8b 44 35 34 41 0f af c5 42 03 44 35 38 42 89 44 35 38 48 8b 0c 24 80 b9 60 03 00 00 00 78 0f 49 8d 0c 2e 48 83 c1 38 31 d2 <f7> 71 f4 89 11 42 8b 7c 35 48 44 01 ef 83 e7 1f 42 89 7c 35 48 48 RSP: 0000:ffff9a0a80108eb0 EFLAGS: 00010046 RAX: 0000000000000000 RBX: 0000000000000019 RCX: ffff90d8c5efc198 RDX: 0000000000000000 RSI: ffff9a0a80549016 RDI: ffff9a0a80549024 RBP: ffff90d8c5efc060 R08: 000000000000197a R09: 00000f604ed00191 R10: 00000000000001e0 R11: ffffffff9468e1d8 R12: 0000000000000020 R13: 0000000000000040 R14: 0000000000000100 R15: 0000000000000002 FS: 00007a75c397aff8(0000) GS:ffff90d912d80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007a77945d1000 CR3: 000000015bf46002 CR4: 0000000000360ea0 Call Trace: <IRQ> __handle_irq_event_percpu+0xa0/0x1c0 handle_irq_event_percpu+0x2d/0x70 handle_irq_event+0x2c/0x48 handle_fasteoi_irq+0xa1/0x161 do_IRQ+0x51/0xd6 common_interrupt+0xf/0xf </IRQ> RIP: 0033:0x7a7856462c59 Code: 89 ca 48 2b 57 20 48 83 c2 10 31 c0 48 3b 57 28 48 0f 46 c1 c3 cc cc cc cc cc cc cc cc cc cc cc cc 64 48 8b 0c 25 00 00 00 00 <b8> f8 02 00 00 48 03 41 08 c3 cc cc cc cc cc cc cc cc cc cc cc cc RSP: 002b:00007a75c39794e8 EFLAGS: 00000246 ORIG_RAX: ffffffffffffffde RAX: 02fa413b24209c6c RBX: 0000017f19e1cf9e RCX: 00007a75c397aff8 RDX: 00007a7855792472 RSI: 00007a7855790aa0 RDI: 0000000000000005 RBP: 0000000000000005 R08: 0000000000000012 R09: 000000000000000d R10: 00000000009f86d2 R11: 000000000000197a R12: 0000017f19e40e7d R13: 000005ee937ae557 R14: 00007a7855790aa0 R15: 00007a7855792472 Modules linked in: ---[ end trace 2ef6d63d0e3d757c ]--- RIP: 0010:snd_intel8x0_interrupt+0x121/0x279 Code: 42 8b 44 35 34 41 0f af c5 42 03 44 35 38 42 89 44 35 38 48 8b 0c 24 80 b9 60 03 00 00 00 78 0f 49 8d 0c 2e 48 83 c1 38 31 d2 <f7> 71 f4 89 11 42 8b 7c 35 48 44 01 ef 83 e7 1f 42 89 7c 35 48 48 RSP: 0000:ffff9a0a80108eb0 EFLAGS: 00010046 RAX: 0000000000000000 RBX: 0000000000000019 RCX: ffff90d8c5efc198 RDX: 0000000000000000 RSI: ffff9a0a80549016 RDI: ffff9a0a80549024 RBP: ffff90d8c5efc060 R08: 000000000000197a R09: 00000f604ed00191 R10: 00000000000001e0 R11: ffffffff9468e1d8 R12: 0000000000000020 R13: 0000000000000040 R14: 0000000000000100 R15: 0000000000000002 FS: 00007a75c397aff8(0000) GS:ffff90d912d80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007a77945d1000 CR3: 000000015bf46002 CR4: 0000000000360ea0 This corresponds to ichdev->position %= ichdev->size; in snd_intel8x0_update(). A print out of that ichdev looks as follows snd_intel8x0 0000:00:18.0: lvi_frag = 0, frags = 0, size = 0, period_size = 0x0, period_size1 = 0x0

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[PATCH] ALSA: intel8x0: Fix breakage at ac97 clock measurement
by Takashi Iwai 08 Jul '21

08 Jul '21

The recent workaround for the wild interrupts in commit c1f0616124c4 ("ALSA: intel8x0: Don't update period unless prepared") leaded to a regression, causing the interrupt storm during ac97 clock measurement at the driver probe. We need to handle the interrupt while the clock measurement as well as the proper PCM streams. Fixes: c1f0616124c4 ("ALSA: intel8x0: Don't update period unless prepared") Reported-and-tested-by: Max Filippov <jcmvbkbc(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAMo8BfKKMQkcsbOQaeEjq_FsJhdK=fn598dvh7YOcZshUSOH… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/intel8x0.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2d1bfbcba933..df3ba5c70de9 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -694,7 +694,7 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich int status, civ, i, step; int ack = 0; - if (!ichdev->prepared || ichdev->suspended) + if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended) return; spin_lock_irqsave(&chip->reg_lock, flags); -- 2.26.2

1 0

[PATCH] ALSA: compress: allow to leave draining state when pausing in draining
by Robert Lee 08 Jul '21

08 Jul '21

When compress offload pauses in draining state, not all platforms need to keep in draining state. Some platforms may call drain or partial drain again when resume from pause in draining, so it needs to wake up from snd_compress_wait_for_drain() in this case. Call API snd_compr_leave_draining_in_pause(), if the platform doesn't need to keep in draining state when pause in draining state. Signed-off-by: Robert Lee <lerobert(a)google.com> --- include/sound/compress_driver.h | 14 ++++++++++++++ sound/core/compress_offload.c | 7 ++++++- 2 files changed, 20 insertions(+), 1 deletion(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 277087f635f3..e16524a93a14 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -145,6 +145,7 @@ struct snd_compr_ops { * @lock: device lock * @device: device id * @use_pause_in_draining: allow pause in draining, true when set + * @leave_draining_in_pause: leave draining state when pausing in draining */ struct snd_compr { const char *name; @@ -156,6 +157,7 @@ struct snd_compr { struct mutex lock; int device; bool use_pause_in_draining; + bool leave_draining_in_pause; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -182,6 +184,18 @@ static inline void snd_compr_use_pause_in_draining(struct snd_compr_stream *subs substream->device->use_pause_in_draining = true; } +/** + * snd_compr_leave_draining_in_pause - Leave draining state when pause in draining + * @substream: compress substream to set + * + * In some platform, we need to leave draining state when we use pause in draining. + * Add API to allow leave draining state. + */ +static inline void snd_compr_leave_draining_in_pause(struct snd_compr_stream *substream) +{ + substream->device->leave_draining_in_pause = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 21ce4c056a92..9c7bd4db6ecd 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -719,8 +719,13 @@ static int snd_compr_pause(struct snd_compr_stream *stream) if (!stream->device->use_pause_in_draining) return -EPERM; retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) + if (!retval) { stream->pause_in_draining = true; + if (stream->device->leave_draining_in_pause) { + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + wake_up(&stream->runtime->sleep); + } + } break; default: return -EPERM; -- 2.32.0.93.g670b81a890-goog

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[PATCH v2] ASoC: atmel: ATMEL drivers depend on HAS_DMA
by Randy Dunlap 08 Jul '21

08 Jul '21

On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" here but several more are needed to prevent kconfig warnings since 'select' does not recognize any dependencies. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Fixes: 3951e4aae2ce ("ASoC: atmel-pcm: dma support based on pcm dmaengine") Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Fixes: 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Cc: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- v2: rebase & resend; drop Cc: to Bo Shen <voice.shen(a)atmel.com> # bounced sound/soc/atmel/Kconfig | 6 ++++++ 1 file changed, 6 insertions(+) --- linux-next-20210628.orig/sound/soc/atmel/Kconfig +++ linux-next-20210628/sound/soc/atmel/Kconfig @@ -19,12 +19,14 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC tristate + depends on HAS_DMA select SND_ATMEL_SOC_DMA select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC_PDC tristate "SoC PCM DAI support for AT91 SSC controller using PDC" depends on ATMEL_SSC + depends on HAS_DMA select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC help @@ -34,6 +36,7 @@ config SND_ATMEL_SOC_SSC_PDC config SND_ATMEL_SOC_SSC_DMA tristate "SoC PCM DAI support for AT91 SSC controller using DMA" depends on ATMEL_SSC + depends on HAS_DMA select SND_ATMEL_SOC_DMA select SND_ATMEL_SOC_SSC help @@ -44,6 +47,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" depends on ARCH_AT91 || COMPILE_TEST depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI + depends on HAS_DMA select SND_ATMEL_SOC_SSC_PDC select SND_SOC_WM8731 help @@ -54,6 +58,7 @@ config SND_ATMEL_SOC_WM8904 tristate "Atmel ASoC driver for boards using WM8904 codec" depends on ARCH_AT91 || COMPILE_TEST depends on ATMEL_SSC && I2C + depends on HAS_DMA select SND_ATMEL_SOC_SSC_DMA select SND_SOC_WM8904 help @@ -64,6 +69,7 @@ config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" depends on ARCH_AT91 || COMPILE_TEST depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI + depends on HAS_DMA select SND_ATMEL_SOC_SSC_DMA select SND_SOC_WM8731 help

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Added configuration information to manual page
by GitHub pull_request - opened 07 Jul '21

07 Jul '21

alsa-project/alsa-utils pull request #103 was opened from braph: Added sections FILES and CONFIGURATION. Removed section BUGS, because the described terminal issues have been fixed. Signed-off-by: braph <braph93(a)gmx.de> Request URL : https://github.com/alsa-project/alsa-utils/pull/103 Patch URL : https://github.com/alsa-project/alsa-utils/pull/103.patch Repository URL: https://github.com/alsa-project/alsa-utils

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Add a note about return values of snd_pcm_state()
by GitHub pull_request - opened 07 Jul '21

07 Jul '21

alsa-project/alsa-lib pull request #162 was opened from HEnquist: This is a suggested improvement to the documentation of `snd_pcm_state()`. There is some confusion, probably caused by old bugs that could made it return a negative error code. This stuff is still floating around the internet and sometimes shows up in searches. This PR adds a comment intended to make it absolutely clear that there is never supposed to be a negative error code. Request URL : https://github.com/alsa-project/alsa-lib/pull/162 Patch URL : https://github.com/alsa-project/alsa-lib/pull/162.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 1/1] ALSA: isa: Fix error return code in snd_cmi8330_probe()
by Zhen Lei 07 Jul '21

07 Jul '21

When 'SB_HW_16' check fails, the error code -ENODEV instead of 0 should be returned, which is the same as that returned when 'WSS_HW_CMI8330' check fails. Fixes: 43bcd973d6d0 ("[ALSA] Add snd_card_set_generic_dev() call to ISA drivers") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/isa/cmi8330.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 6c59a3eec59b..ef6d0a20efd8 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -551,7 +551,7 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) } if (acard->sb->hardware != SB_HW_16) { snd_printk(KERN_ERR PFX "SB16 not found during probe\n"); - return err; + return -ENODEV; } snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */ -- 2.25.1

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[PATCH] ALSA: usb-audio: Reduce latency at playback start, take#2
by Takashi Iwai 07 Jul '21

07 Jul '21

This is another attempt for the reduction of the latency at the start of a USB audio playback stream. The first attempt in the commit 9ce650a75a3b caused an unexpected regression (a deadlock with pipewire usage) and was later reverted by the commit 4b820e167bf6. The devils are always living in details, of course; the cause of the deadlock was the call of snd_pcm_period_elapsed() inside prepare_playback_urb() callback. In the original code, this callback is never called from the stream lock context as it's driven solely from the URB complete callback. Along with the movement of the URB submission into the trigger START, this prepare call may be also executed in the stream lock context, hence it deadlocked with the another lock in snd_pcm_period_elapsed(). (Note that this happens only conditionally with a small period size that matches with the URB buffer length, which was a reason I overlooked during my tests. Also, the problem wasn't seen in the capture stream because the capture stream handles the period-elapsed only at retire callback that isn't executed at the trigger.) If it were only about avoiding the deadlock, it'd be possible to use snd_pcm_period_elapsed_under_stream_lock() as a solution. However, in general, the period elapsed notification must be sent after the actual stream start, and replacing the call wouldn't satisfy the pattern. A better option is to delay the notification after the stream start procedure finished, instead. In the case of USB framework, one of the fitting place would be the complete callback of the first URB. So, as a workaround of the deadlock and the order fixes above, in addition to the re-applying the changes in the commit 9ce650a75a3, this patch introduces a new flag indicating the delayed period-elapsed handling and sets it under the possible deadlock condition (i.e. prepare callback being called before subs->running is set). Once when the flag is set, the period-elapsed call is handled at a later URB complete call instead. As a reference for the original motivation for the low-latency change, I cite here again: | USB-audio driver behaves a bit strangely for the playback stream -- | namely, it starts sending silent packets at PCM prepare state while | the actual data is submitted at first when the trigger START is | kicked off. This is a workaround for the behavior where URBs are | processed too quickly at the beginning. That is, if we start | submitting URBs at trigger START, the first few URBs will be | immediately completed, and this would result in the immediate | period-elapsed calls right after the start, which may confuse | applications. | | OTOH, submitting the data after silent URBs would, of course, result | in a certain delay of the actual data processing, and this is rather | more serious problem on modern systems, in practice. | | This patch tries to revert the workaround and lets the URB | submission starting at PCM trigger for the playback again. As far | as I've tested with various backends (native ALSA, PA, JACK, PW), I | haven't seen any problems (famous last words :) | | Note that the capture stream handling needs no such workaround, | since the capture is driven per received URB. Link: https://lore.kernel.org/r/4e71531f-4535-fd46-040e-506a3c256bbd@marcan.st Link: https://lore.kernel.org/r/s5hbl7li0fe.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.h | 1 + sound/usb/pcm.c | 26 +++++++++++++++++++++----- 2 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index f41dbdc31336..6c0a052a28f9 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -158,6 +158,7 @@ struct snd_usb_substream { unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */ unsigned int running: 1; /* running status */ + unsigned int period_elapsed_pending; /* delay period handling */ unsigned int buffer_bytes; /* buffer size in bytes */ unsigned int inflight_bytes; /* in-flight data bytes on buffer (for playback) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e26d37365f02..4e5031a68064 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -611,13 +611,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->hwptr_done = 0; subs->transfer_done = 0; subs->last_frame_number = 0; + subs->period_elapsed_pending = 0; runtime->delay = 0; - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) - ret = start_endpoints(subs); - unlock: snd_usb_unlock_shutdown(chip); return ret; @@ -1398,6 +1394,10 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->trigger_tstamp_pending_update = false; } + if (period_elapsed && !subs->running) { + subs->period_elapsed_pending = 1; + period_elapsed = 0; + } spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -1413,6 +1413,7 @@ static void retire_playback_urb(struct snd_usb_substream *subs, { unsigned long flags; struct snd_urb_ctx *ctx = urb->context; + bool period_elapsed = false; spin_lock_irqsave(&subs->lock, flags); if (ctx->queued) { @@ -1423,13 +1424,20 @@ static void retire_playback_urb(struct snd_usb_substream *subs, } subs->last_frame_number = usb_get_current_frame_number(subs->dev); + if (subs->running) { + period_elapsed = subs->period_elapsed_pending; + subs->period_elapsed_pending = 0; + } spin_unlock_irqrestore(&subs->lock, flags); + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); } static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_usb_substream *subs = substream->runtime->private_data; + int err; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -1440,6 +1448,14 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea prepare_playback_urb, retire_playback_urb, subs); + if (cmd == SNDRV_PCM_TRIGGER_START) { + err = start_endpoints(subs); + if (err < 0) { + snd_usb_endpoint_set_callback(subs->data_endpoint, + NULL, NULL, NULL); + return err; + } + } subs->running = 1; dev_dbg(&subs->dev->dev, "%d:%d Start Playback PCM\n", subs->cur_audiofmt->iface, -- 2.26.2

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