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

KMSAN: uninit-value in snd_seq_oss_synth_make_info (2)
by syzbot 15 Jan '21

15 Jan '21

Hello, syzbot found the following issue on: HEAD commit: 73d62e81 kmsan: random: prevent boot-time reports in _mix_.. git tree: https://github.com/google/kmsan.git master console output: https://syzkaller.appspot.com/x/log.txt?x=10284200d00000 kernel config: https://syzkaller.appspot.com/x/.config?x=2cdf4151c9653e32 dashboard link: https://syzkaller.appspot.com/bug?extid=e42504ff21cff05a595f compiler: clang version 11.0.0 (https://github.com/llvm/llvm-project.git ca2dcbd030eadbf0aa9b660efe864ff08af6e18b) userspace arch: i386 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+e42504ff21cff05a595f(a)syzkaller.appspotmail.com ===================================================== BUG: KMSAN: uninit-value in strlen lib/string.c:576 [inline] BUG: KMSAN: uninit-value in strlcpy+0x8c/0x1c0 lib/string.c:143 CPU: 1 PID: 10203 Comm: syz-executor.3 Not tainted 5.10.0-rc4-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x21c/0x280 lib/dump_stack.c:118 kmsan_report+0xf7/0x1e0 mm/kmsan/kmsan_report.c:118 __msan_warning+0x5f/0xa0 mm/kmsan/kmsan_instr.c:197 strlen lib/string.c:576 [inline] strlcpy+0x8c/0x1c0 lib/string.c:143 snd_seq_oss_synth_make_info+0x381/0x820 sound/core/seq/oss/seq_oss_synth.c:619 snd_seq_oss_synth_info_user sound/core/seq/oss/seq_oss_ioctl.c:24 [inline] snd_seq_oss_ioctl+0x12b4/0x2530 sound/core/seq/oss/seq_oss_ioctl.c:139 odev_ioctl sound/core/seq/oss/seq_oss.c:180 [inline] odev_ioctl_compat+0x247/0x290 sound/core/seq/oss/seq_oss.c:190 __do_compat_sys_ioctl fs/ioctl.c:842 [inline] __se_compat_sys_ioctl+0x53d/0x1100 fs/ioctl.c:793 __ia32_compat_sys_ioctl+0x4a/0x70 fs/ioctl.c:793 do_syscall_32_irqs_on arch/x86/entry/common.c:80 [inline] __do_fast_syscall_32+0x102/0x160 arch/x86/entry/common.c:139 do_fast_syscall_32+0x6a/0xc0 arch/x86/entry/common.c:162 do_SYSENTER_32+0x73/0x90 arch/x86/entry/common.c:205 entry_SYSENTER_compat_after_hwframe+0x4d/0x5c RIP: 0023:0xf7f8c549 Code: b8 01 10 06 03 74 b4 01 10 07 03 74 b0 01 10 08 03 74 d8 01 00 00 00 00 00 00 00 00 00 00 00 00 00 51 52 55 89 e5 0f 34 cd 80 <5d> 5a 59 c3 90 90 90 90 eb 0d 90 90 90 90 90 90 90 90 90 90 90 90 RSP: 002b:00000000f55860cc EFLAGS: 00000296 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000004 RCX: 00000000c08c5114 RDX: 0000000020000080 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Local variable ----minf@snd_seq_oss_synth_make_info created at: snd_seq_oss_synth_make_info+0x22a/0x820 sound/core/seq/oss/seq_oss_synth.c:613 snd_seq_oss_synth_make_info+0x22a/0x820 sound/core/seq/oss/seq_oss_synth.c:613 ===================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

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Question about daifmt of legacy DT on simple-card
by Kuninori Morimoto 15 Jan '21

15 Jan '21

Hi ALSA SoC Cc Jyri For historical reason, simple-card supports legacy DT daifmt settings. Then, I noticed one strange code. -- simple-card-utils.c --- asoc_simple_parse_daifmt(xxx) { (A) daifmt = snd_soc_of_parse_daifmt(...); daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; if (!bitclkmaster && !framemaster) { /* * No dai-link level and master setting was not found from * sound node level, revert back to legacy DT parsing and * take the settings from codec node. */ (B) daifmt = snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) | (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); } ... ... } It rollbacks to legacy DT parsing at (B) if (A) didn't have master settings. Here, (B) re-try to get daifmt, and use "or" with (daifmt & ~CLOCK mask). Why CLOCK mask ? and shouldn't it use mask when "or" ? Otherwise FORMAT and INV part will be duplicated, I think. for example daifmt = (snd_soc_of_parse_daifmt() & SND_SOC_DAIFMT_CLOCK_MASK) | (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK) I think using snd_soc_of_parse_daifmt() only is very enough at (B), but am I misunderstanding ?? The original code was from b3ca11ff59bc5842b01f13421a17e6d9a8936784 ("ASoC: simple-card: Move dai-link level properties away from dai subnodes") Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 1/2] [v2] ALSA: hda: fix SND_INTEL_DSP_CONFIG dependency
by Arnd Bergmann 14 Jan '21

14 Jan '21

From: Arnd Bergmann <arnd(a)arndb.de> The sof-pci-dev driver fails to link when built into the kernel and CONFIG_SND_INTEL_DSP_CONFIG is set to =m: arm-linux-gnueabi-ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': sof-pci-dev.c:(.text+0x1c): undefined reference to `snd_intel_dsp_driver_probe' All other drivers using this interface already use a 'select SND_INTEL_DSP_CONFIG' statement to force the it to be present, so it seems reasonable to do the same here. The stub implementation in the header makes the problem harder to find, as it avoids the link error when SND_INTEL_DSP_CONFIG is completely disabled, without any obvious upsides. Remove these stubs to make it clearer that the driver is in fact needed here. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: fix various build issues in the first version, now passes all randconfig builds I tried --- include/sound/intel-dsp-config.h | 17 ----------------- sound/soc/sof/Kconfig | 2 ++ sound/soc/sof/intel/Kconfig | 4 ++-- sound/soc/sof/intel/byt.c | 2 +- 4 files changed, 5 insertions(+), 20 deletions(-) diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h index d4609077c258..94667e870029 100644 --- a/include/sound/intel-dsp-config.h +++ b/include/sound/intel-dsp-config.h @@ -18,24 +18,7 @@ enum { SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF }; -#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) - int snd_intel_dsp_driver_probe(struct pci_dev *pci); int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]); -#else - -static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -static inline -int snd_intel_acpi_dsp_driver_probe(struct device *dev, const u8 acpi_hid[ACPI_ID_LEN]) -{ - return SND_INTEL_DSP_DRIVER_ANY; -} - -#endif - #endif diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..051fd3d27047 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -12,6 +12,7 @@ if SND_SOC_SOF_TOPLEVEL config SND_SOC_SOF_PCI tristate "SOF PCI enumeration support" depends on PCI + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI help @@ -23,6 +24,7 @@ config SND_SOC_SOF_PCI config SND_SOC_SOF_ACPI tristate "SOF ACPI enumeration support" depends on ACPI || COMPILE_TEST + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF select SND_SOC_ACPI if ACPI select IOSF_MBI if X86 && PCI diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index ff9266413a06..67365ce0d86d 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_SOF_INTEL_COMMON if SND_SOC_SOF_ACPI config SND_SOC_SOF_BAYTRAIL - bool "SOF support for Baytrail, Braswell and Cherrytrail" + tristate "SOF support for Baytrail, Braswell and Cherrytrail" select SND_SOC_SOF_INTEL_ATOM_HIFI_EP select SND_INTEL_DSP_CONFIG help @@ -70,7 +70,7 @@ config SND_SOC_SOF_BAYTRAIL If unsure select "N". config SND_SOC_SOF_BROADWELL - bool "SOF support for Broadwell" + tristate "SOF support for Broadwell" select SND_INTEL_DSP_CONFIG select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_INTEL_HIFI_EP_IPC diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 65abcca94521..42dba3da1ca3 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -660,7 +660,7 @@ EXPORT_SYMBOL_NS(tng_chip_info, SND_SOC_SOF_MERRIFIELD); #endif /* CONFIG_SND_SOC_SOF_MERRIFIELD */ -#ifdef CONFIG_SND_SOC_SOF_BAYTRAIL +#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) static void byt_reset_dsp_disable_int(struct snd_sof_dev *sdev) { /* Disable Interrupt from both sides */ -- 2.29.2

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[PATCH v2 0/2] Enable DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 14 Jan '21

14 Jan '21

v2: Update patch to align with latest kernel release. v1: Initial patch version, to enable DMA mode on Intel Keembay platform. Michael Sit Wei Hong (2): dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 157 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + 4 files changed, 167 insertions(+), 15 deletions(-) -- 2.17.1

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USB-Audio/ALC1220: Bump analog Speaker priority over S/PDIF
by GitHub pull_request - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-ucm-conf pull request #73 was opened from MarijnS95: Suggestion from https://github.com/alsa-project/alsa-ucm-conf/pull/25#issuecomment-760141633 The `S/PDIF` port does not have any jack sensing, and with a priority higher than the `Speakers` it will always be selected by default instead of the `Speakers` even if unplugged. Swapping the priorities around allows analog `Speakers` to be selected first, _if_ they are plugged in. Otherwise `S/PDIF` is used. The sections are reordered to adhere to this priority, including some indentation fixes. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/73 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/73.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/2] ALSA: Simplify with dma_set_mask_and_coherent()
by Takashi Iwai 14 Jan '21

14 Jan '21

Simple patches that convert the two calls of dma_set_mask*() into the single dma_set_mask_and_coherent() helper. Just a minor cleanup, there should be no functional changes. Takashi === Takashi Iwai (2): ALSA: pci: Simplify with dma_set_mask_and_coherent() ALSA: x86: Simplify with dma_set_mask_and_coherent() sound/pci/ad1889.c | 3 +-- sound/pci/ali5451/ali5451.c | 3 +-- sound/pci/als300.c | 3 +-- sound/pci/als4000.c | 3 +-- sound/pci/au88x0/au88x0.c | 3 +-- sound/pci/aw2/aw2-alsa.c | 3 +-- sound/pci/azt3328.c | 3 +-- sound/pci/ca0106/ca0106_main.c | 3 +-- sound/pci/cs5535audio/cs5535audio.c | 3 +-- sound/pci/ctxfi/cthw20k1.c | 8 ++------ sound/pci/ctxfi/cthw20k2.c | 8 ++------ sound/pci/es1938.c | 3 +-- sound/pci/es1968.c | 3 +-- sound/pci/hda/hda_intel.c | 8 ++------ sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/maestro3.c | 3 +-- sound/pci/sonicvibes.c | 3 +-- sound/pci/trident/trident_main.c | 3 +-- sound/x86/intel_hdmi_audio.c | 3 +-- 19 files changed, 22 insertions(+), 50 deletions(-) -- 2.26.2

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[PATCH] ALSA: hda/realtek - Limit int mic boost on Acer Aspire E5-575T
by Chris Chiu 14 Jan '21

14 Jan '21

The Acer Apire E5-575T laptop with codec ALC255 has a terrible background noise comes from internal mic capture. And the jack sensing dose not work for headset like some other Acer laptops. This patch limits the internal mic boost on top of the existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c1d2a3fb1a4..60eb8383a704 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6371,6 +6371,7 @@ enum { ALC256_FIXUP_HP_HEADSET_MIC, ALC236_FIXUP_DELL_AIO_HEADSET_MIC, ALC282_FIXUP_ACER_DISABLE_LINEOUT, + ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7808,6 +7809,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7826,6 +7833,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), -- 2.20.1

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[PATCH v6 1/2] cros_ec_commands: Add EC_CODEC_I2S_RX_RESET
by Yu-Hsuan Hsu 14 Jan '21

14 Jan '21

Add the new command EC_CODEC_I2S_RX_RESET in ec_codec_i2s_rx_subcmd, which is used for resetting the EC codec. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- include/linux/platform_data/cros_ec_commands.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 86376779ab31..95889ada83a3 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -4600,6 +4600,7 @@ enum ec_codec_i2s_rx_subcmd { EC_CODEC_I2S_RX_SET_SAMPLE_DEPTH = 0x2, EC_CODEC_I2S_RX_SET_DAIFMT = 0x3, EC_CODEC_I2S_RX_SET_BCLK = 0x4, + EC_CODEC_I2S_RX_RESET = 0x5, EC_CODEC_I2S_RX_SUBCMD_COUNT, }; -- 2.30.0.284.gd98b1dd5eaa7-goog

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ucm2: add initial configuration for TRX40 Gigabyte Aorus Master Audio
by GitHub pull_request - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-ucm-conf pull request #72 was opened from perexg: BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/25 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/72 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/72.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Audio over HDMI via USB-C Dock broke recently - module-alsa-card.c: Failed to find a working profile
by GitHub issues - opened 14 Jan '21

14 Jan '21

alsa-project/alsa-lib issue #112 was opened from asgrim: Hi folks, not sure if this is related to #104 since I get the same error, but slightly different characteristics. Running Ubuntu 20.04.1 here. I've got an XPS 17 9700, which is plugged into a Dell WD19TB dock via USB-C. Off that, I have a Dell 34" screen via HDMI, into which my speakers are plugged in. My headphones are plugged directly into the WD19TB dock. I do have a Yeti mic as well, which is plugged via USB into the dock also. I often switch between my speakers/headphones using a Gnome extension (I select either `Headphones (USB Audio)` or HDMI something (can't remember what it was called exactly). However, I noticed recently (maybe last week or two?) I wasn't able to switch. Looked into it today, it's because the HDMI audio output seems completely missing. Does not show in regular Ubuntu sound settings, nor in `pavucontrol`. Checked /var/log/syslog, and found this `Failed to find a working profile` message. Some Googling bought me here, hopefully you lovely folks can help! * `pacmd ls` output is [here](https://gist.github.com/asgrim/1788fcae11cd1d0ced106c5d94199f29#file-… * `pulseaudio -vvvvv` output is [here](https://gist.github.com/asgrim/1788fcae11cd1d0ced106c5d94199f29#file-… Hopefully this is just user error, but this did used to work, and I don't know where to ask for help as I know nothing about Linux audio stuff :D Side note: I'm aware there's an issue with my XPS 17 "onboard" speakers not working, they've never worked; I believe there's a fix out there, but it looks complicated so I have so far avoided it. I'm ok with this as I've got the external speakers anyway. Issue URL : https://github.com/alsa-project/alsa-lib/issues/112 Repository URL: https://github.com/alsa-project/alsa-lib

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