June 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: ac97: Fix possible NULL dereference in snd_ac97_mixer
by Su Hui 23 Aug '23

23 Aug '23

smatch error: sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error: we previously assumed 'rac97' could be null (see line 2072) remove redundant assignment, return error if rac97 is NULL. Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 9afc5906d662..80a65b8ad7b9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2069,8 +2069,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, .dev_disconnect = snd_ac97_dev_disconnect, }; - if (rac97) - *rac97 = NULL; + if (!rac97) + return -EINVAL; if (snd_BUG_ON(!bus || !template)) return -EINVAL; if (snd_BUG_ON(template->num >= 4)) -- 2.30.2

3 4

[PATCH 0/4] ASoC: amd: acp: Add sound support for a line of HUAWEI laptops
by Marian Postevca 21 Aug '23

21 Aug '23

This series adds support for a line of HUAWEI laptops with AMD CPUs that connect using the ACP3x module to a ES8336 codec. The codec driver must be extended to support the S32 LE format and the MCLK div by 2 option. MCLK div by 2 is needed for one specific SKU which uses a 48Mhz MCLK which seems to be too high of a frequency for the codec and must be divided by 2. The acp legacy driver must also be extended by using callbacks so that the more complicated handling for this specific codec can be moved outside the more generic ACP code. The last patch tries to avoid anoying pop sounds when the speaker/headphones are enabled/disabled by delaying the handling of the GPIOs and using a mutex to avoid race conditions between the speaker power event callback and the trigger callback. Marian Postevca (4): ASoC: es8316: Enable support for S32 LE format and MCLK div by 2 ASoC: amd: acp: Add support for splitting the codec specific code from the ACP driver ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec ASoC: amd: acp: Improve support for speaker power events sound/soc/amd/acp-config.c | 70 ++ sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-legacy-mach.c | 105 ++- sound/soc/amd/acp/acp-mach-common.c | 8 + sound/soc/amd/acp/acp-mach.h | 67 ++ sound/soc/amd/acp/acp-renoir.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 615 ++++++++++++++++++ sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h | 12 + sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8316.h | 3 + 10 files changed, 886 insertions(+), 21 deletions(-) create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h -- 2.39.1

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Fwd: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot
by Joseph C. Sible 16 Aug '23

16 Aug '23

Hello, Forwarding my regression bug report submission per the instructions in "Reporting regressions". #regzbot introduced: 5aec989130 https://bugzilla.kernel.org/show_bug.cgi?id=217440 Regards, Joseph C. Sible ---------- Forwarded message --------- From: <bugzilla-daemon(a)kernel.org> Date: Sat, May 13, 2023 at 3:46 PM Subject: [Bug 217440] New: ALC236 audio disappears from HP 15z-fc000 on warm boot To: <josephcsible(a)gmail.com> https://bugzilla.kernel.org/show_bug.cgi?id=217440 Bug ID: 217440 Summary: ALC236 audio disappears from HP 15z-fc000 on warm boot Product: Drivers Version: 2.5 Kernel Version: 6.3.2 Hardware: All OS: Linux Status: NEW Severity: normal Priority: P3 Component: Sound(ALSA) Assignee: perex(a)perex.cz Reporter: josephcsible(a)gmail.com Regression: Yes Bisected 5aec98913095ed3b4424ed6c5fdeb6964e9734da commit-id: On an HP 15z-fc000 laptop that uses Realtek ALC236 for sound, once the system is warm booted ("reboot" from within Linux), sound stops working completely and the device doesn't even show up anymore. Once this happens, the only way to get it to come back is to cold boot ("poweroff" from within Linux, wait for it to turn completely off, and turn it back on with the physical power button). In particular, once you boot from a "bad" kernel, no number of warm boots with "good" kernels will ever bring sound back. I bisected this to commit 5aec989, which first appeared in 5.15-rc6. In addition, I confirmed by doing a custom build of 6.3.2 with reverts of both f30741c (just to avoid a conflict) and 5aec989, which works correctly for me. -- You may reply to this email to add a comment. You are receiving this mail because: You reported the bug.

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Audio output heavily distorted on Xen PV dom0 if not recording (Intel ADL)
by Marek Marczykowski-Górecki 01 Aug '23

01 Aug '23

Hi, I have a rather interesting issue on HP EliteBook 1040, with Intel ADL HDA. Sometimes (often) audio output is heavily distorted, but then opening pavucontrol or just launching `parecord /dev/null` fixes the issue instantly. But, when the mixer or recording is closed, then few seconds later issue comes back. You can hear this at https://cloud.marmarek.net/s/46prqZnwrqDkBdD Initially recording was running, then about 10s stopped and you can hear the issue few seconds later. Then I starter recording again, and the issue is gone. This is how it should sound: https://youtube.com/watch?v=4Tr0otuiQuU I do not see any kernel messages when issue happens. The issue happens on Linux 6.1.26, 6.2.10, and 6.3.2. I haven't tested other versions. It happens only when running under Xen (in PV dom0). The very same Linux started directly works fine. Full dmesg of both native and Xen boots is at https://gist.github.com/marmarek/548ee79282446344a172109bcd943930 When running Linux 6.3.2 there, I noticed that just after boot first playback is okay, but if I start and stop recording it breaks. And then after some time of idle it fixes itself again. I have not observed this behavior on 6.2.10 (it was broken initially too), but I could be also doing some things differently. I have few other ADL-based systems (but not HP), and they do not show this issue, only this HP laptop is affected. Example dmesg from another ADL-based system: https://gist.github.com/marmarek/38c5ba3ec58f9300fbc1842ddc9fa4f5 I'm not sure if that's the relevant difference, but comparing dmesg from working and not working systems, I see those on boot for the problematic case: snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: enabling device (0000 -> 0002) xen: registering gsi 16 triggering 0 polarity 1 Already setup the GSI :16 sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) sof-audio-pci-intel-tgl 0000:00:1f.3: use msi interrupt mode sof-audio-pci-intel-tgl 0000:00:1f.3: hda codecs found, mask 5 sof-audio-pci-intel-tgl 0000:00:1f.3: using HDA machine driver skl_hda_dsp_generic now sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: unknown sof_ext_man header type 3 size 0x30 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:0 skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Parent card not yet available, widget card binding deferred cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.0 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.1 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) Any ideas? -- Best Regards, Marek Marczykowski-Górecki Invisible Things Lab

3 6

[PATCH 00/11] Even more msm bindings fixes
by Konrad Dybcio 25 Jul '23

25 Jul '23

This series contains couple more random dt schema warning fixes, centered around linux-msm. All of the patches (except) the last two are pretty much independent. Patch 1 is not only a bindings fix, but also a functional one. Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (11): arm64: dts: qcom: sc7180: Fix DSI0_PHY reg-names arm64: dts: qcom: msm8939: Drop "qcom,idle-state-spc" compatible arm64: dts: qcom: msm8939: Add missing 'cache-unified' to L2 arm64: dts: qcom: apq8039-t2: Drop inexistent property dt-bindings: sound: Convert pm8916-wcd-analog-codec to YAML dt-bindings: mfd: qcom,spmi-pmic: Reference pm8916 wcd analog codec schema dt-bindings: iommu: qcom_iommu: Allow 'tbu' clock dt-bindings: usb: ci-hdrc-usb2: Fix handling pinctrl properties dt-bindings: arm: msm: kpss-acc: Make the optional reg truly optional dt-bindings: usb: dwc3: Allow just 1 interrupt for MSM8996 arm64: dts: qcom: msm8996: Add missing interrupt to the USB2 controller .../devicetree/bindings/arm/msm/qcom,kpss-acc.yaml | 1 + .../devicetree/bindings/iommu/qcom,iommu.yaml | 4 + .../devicetree/bindings/mfd/qcom,spmi-pmic.yaml | 2 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 101 ------------- .../sound/qcom,pm8916-wcd-analog-codec.yaml | 160 +++++++++++++++++++++ .../devicetree/bindings/usb/ci-hdrc-usb2.yaml | 27 +--- .../devicetree/bindings/usb/qcom,dwc3.yaml | 2 +- arch/arm64/boot/dts/qcom/apq8039-t2.dts | 1 - arch/arm64/boot/dts/qcom/msm8939.dtsi | 4 +- arch/arm64/boot/dts/qcom/msm8996.dtsi | 3 + arch/arm64/boot/dts/qcom/sc7180.dtsi | 4 +- 11 files changed, 182 insertions(+), 127 deletions(-) --- base-commit: 3eedd211ad93c322fb360b83a3d76a2c6cd622dc change-id: 20230627-topic-more_bindings-325a6f4b93a7 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

9 29

[PATCH v1 0/5] Add PWM-DAC audio support for StarFive JH7110 RISC-V SoC
by Hal Feng 25 Jul '23

25 Jul '23

This patchset adds PWM-DAC audio support for the StarFive JH7110 SoC. The PWM-DAC module does not require a hardware codec, so add a dummy codec driver for it. The fourth patch depends on the patch [1]. The fifth patch depends on the patchset [2], patch [3]. [1] https://lore.kernel.org/all/20230526145402.450-3-walker.chen@starfivetech.c… [2] https://lore.kernel.org/all/20230518101234.143748-1-xingyu.wu@starfivetech.… [3] https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Hal Feng (5): ASoC: dt-bindings: Add StarFive JH7110 dummy PWM-DAC transmitter ASoC: codecs: Add StarFive JH7110 dummy PWM-DAC transmitter driver ASoC: dt-bindings: Add StarFive JH7110 PWM-DAC controller ASoC: starfive: Add JH7110 PWM-DAC driver riscv: dts: starfive: Add JH7110 PWM-DAC support .../sound/starfive,jh7110-pwmdac-dit.yaml | 38 + .../sound/starfive,jh7110-pwmdac.yaml | 76 ++ MAINTAINERS | 8 + .../jh7110-starfive-visionfive-2.dtsi | 50 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 13 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jh7110_pwmdac_transmitter.c | 74 ++ sound/soc/starfive/Kconfig | 9 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh7110_pwmdac.c | 787 ++++++++++++++++++ 11 files changed, 1062 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac-dit.yaml create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml create mode 100644 sound/soc/codecs/jh7110_pwmdac_transmitter.c create mode 100644 sound/soc/starfive/jh7110_pwmdac.c base-commit: 45a3e24f65e90a047bef86f927ebdc4c710edaa1 prerequisite-patch-id: 8c735dffc6d5388a35a76b16e914a2f9722ad979 prerequisite-patch-id: ffa1f5831e75722c9f41603f009f762c9fd525e2 prerequisite-patch-id: 36e69700dfc0375b950b0e23086ed3b722cb84a4 prerequisite-patch-id: 0b49b996d7a404ea548e1734c12933ec749e92b9 prerequisite-patch-id: 81f7c65712c4901a7a178ddcd98ffc55f3b473ff prerequisite-patch-id: f342fbf594014b072378528bea94c01fb2186e1a prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: d5abfba63fc07ff97b5023911513c260bb7a53e1 prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 prerequisite-patch-id: 2f7aca99e714a4c590a91baa015080ac0902814d prerequisite-patch-id: 32cabbc4e7a97ec14d5c28a477fa483784f86709 prerequisite-patch-id: d449b1957dd77c2537c38585daa75974c94c529a -- 2.38.1

3 20

[PATCH v5 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 21 Jul '23

21 Jul '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1189 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 281 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 969 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7636 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

2 11

[PATCH 0/6] Add audio support in v4l2 framework
by Shengjiu Wang 19 Jul '23

19 Jul '23

Audio signal processing has the requirement for memory to memory similar as Video. This patch is to add this support in v4l2 framework, defined new buffer type V4L2_BUF_TYPE_AUDIO_CAPTURE and V4L2_BUF_TYPE_AUDIO_OUTPUT, defined new format v4l2_audio_format for audio case usage. The created audio device is named "/dev/audioX". And add memory to memory support for two kinds of i.MX ASRC module Shengjiu Wang (6): media: v4l2: Add audio capture and output support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: Add memory to memory driver ASoC: fsl_asrc: enable memory to memory function ASoC: fsl_easrc: enable memory to memory function .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/v4l2-core/v4l2-dev.c | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 52 ++ include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + include/uapi/linux/videodev2.h | 19 + sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 175 +++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 54 ++ sound/soc/fsl/fsl_asrc_m2m.c | 878 ++++++++++++++++++ sound/soc/fsl/fsl_asrc_m2m.h | 48 + sound/soc/fsl/fsl_easrc.c | 255 ++++- sound/soc/fsl/fsl_easrc.h | 6 + 15 files changed, 1557 insertions(+), 4 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c create mode 100644 sound/soc/fsl/fsl_asrc_m2m.h -- 2.34.1

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[PATCH v2 0/8] ALSA: emu10k1: add support for high-bitrate modes of E-MU cards
by Oswald Buddenhagen 17 Jul '23

17 Jul '23

This series is what all the work was about: support the "dual-/quad-pumped" modes of the E-MU cards. Oswald Buddenhagen (8): ALSA: emu10k1: introduce alternative E-MU D.A.S. mode ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode ALSA: emu10k1: set the "no filtering" bits on PCM voices ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit ALSA: add snd_ctl_add_locked() ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode include/sound/control.h | 1 + include/sound/emu10k1.h | 11 + sound/core/control.c | 31 ++ sound/pci/emu10k1/emu10k1.c | 29 +- sound/pci/emu10k1/emu10k1_main.c | 19 +- sound/pci/emu10k1/emufx.c | 109 +++- sound/pci/emu10k1/emumixer.c | 901 +++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 422 +++++++++++++-- sound/pci/emu10k1/emuproc.c | 5 + sound/pci/emu10k1/io.c | 30 +- sound/pci/emu10k1/voice.c | 6 + 11 files changed, 1383 insertions(+), 181 deletions(-) Range-diff against v1: 1: c8c1fbba5e52 = 1: 7cb49147164f ALSA: emu10k1: introduce alternative E-MU D.A.S. mode 2: 879066428bee = 2: aa9ead00e045 ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode 3: b663229a0f46 = 3: ac830e67322f ALSA: emu10k1: set the "no filtering" bits on PCM voices 4: a9de9a73571e = 4: 26e0b7c02c1d ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit 5: 0f1950b03193 ! 5: 94b64e6c123d ALSA: add snd_ctl_add_locked() @@ Commit message This is in fact more symmetrical to snd_ctl_remove() than snd_ctl_add() is - the former could be named snd_ctl_remove_locked() just as well. + One may argue that this is going in the wrong direction, as drivers have + no business managing the lock. This may be true in principle, but in + practice the vast majority of controls is created even before the device + was registered, and therefore before any locking is necessary at all. + That means that an even more radical approach of changing snd_ctl_add() + do do no locking, and converting the call sites that actually need + locking to a new function, would better match reality, and would be + somewhat more efficient at that. However, that seems a bit risky and way + too much work. + This will be used to dynamically change the available controls from another control's put() callback, which is already locked. One might want to add snd_ctl_replace_locked() for completeness, but I have no use for it now. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - extended commit message ## include/sound/control.h ## @@ include/sound/control.h: void snd_ctl_notify_one(struct snd_card * card, unsigned int mask, struct snd_kc 6: 511efe4ac7ad ! 6: 7bc314bae7f2 ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode @@ Commit message on playback, and throw away the excess ones on capture. Input-to-output monitoring does actually use the full sample rate, though. - Notably, add_ctls() now uses snd_ctl_add_locked(), so it doesn't - deadlock when called from snd_emu1010_clock_shift_put(). This also - affects the initial creation of the controls, which is OK, as that is - done before the card is registered, so no concurrent access can occur. + Due to hardware constraints, changing the clock multiplier (CM) changes + the available audio ports and the number of available channels. This has + an impact on the channel routing mixer controls. One way to deal with + this would be presenting a union of all possibilities, and simply + ignoring currently inapplicable settings. However, this would be a + terrible user experience, and go against the spirit of prior patches + aimed at decluttering the mixer. Therefore, we do dynamic + reconfiguration (DR) of the mixer in response to changing the CM. + + DR is somewhat controversial, as it has the potential to crash poorly + programmed applications. But that in itself isn't a very convincing + argument against it, as by that logic we'd have to ban all hot-plugging. + Such crashes would also not really qualify as regressions, as the D.A.S. + mode is a new opt-in feature, and therefore no previously stable setups + would be impacted. Also, pendantically, the driver already had DR via + SNDRV_EMU10K1_IOCTL_CODE_POKE. A somewhat valid concern is that changing + mixer settings is a non-privileged operation and therefore potential + crashes could be exploited for a somewhat more impactful nuisance attack + on another user than messing with the mixer per se. However, systemd & + co. limit device access to the user currently logged in on the seat + owning the device. + + There is a specific concern about doing DR in response to changing a + mixer control's value, as an application may legitimately react to DR by + updating all mixer settings in turn. However, that update should write + the same value to the clock multiplier, thus terminating the recursion. + + One may limit DR to merely disabling inapplicable controls, in the hope + that this would be better handled than completely tearing down and + re-creating controls as we do. However, there is no guarantee for that. + And because it is impossible to disable particular enum values within a + control, it would be necessary to have three complete sets of + per-channel controls. This would yield an extremely cluttered and + confusing UI if the application (reasonably) chose to merely visually + disable inactive controls rather than hiding them. + + We do the DR synchronously from within snd_emu1010_clock_shift_put(). + This was enabled by commit 5bbb1ab5bd ("control: use counting semaphore + as write lock for ELEM_WRITE operation"); we merely need to make + add_ctls() use snd_ctl_add_locked() instead of snd_ctl_add(), so it + doesn't deadlock. That also affects the initial creation of the + controls, which is OK, as that is done before the card is registered, so + no concurrent access can occur. + + It would be possible to do the DR in a tasklet after the ioctl finishes. + However, it is not obvious what actual problem that would solve, and the + added asynchronicity would significantly complicate matters, esp. wrt. + the batch updates expected during mixer state restoration. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - expanded commit message ## include/sound/emu10k1.h ## @@ include/sound/emu10k1.h: struct snd_emu1010 { 7: d5cb50ca707f = 7: 72a156fb32cd ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode 8: 319425a4ccb6 ! 8: 6d35891832b3 ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode @@ Commit message d948035a92 ("Remove PCM xfer_align sw params"). Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> + --- + v2: + - fixed `sparse` warning re missing __user annotation ## sound/pci/emu10k1/emufx.c ## @@ sound/pci/emu10k1/emufx.c: static int _snd_emu10k1_das_init_efx(struct snd_emu10k1 *emu) @@ sound/pci/emu10k1/emupcm.c: static int snd_emu10k1_efx_playback_trigger(struct s + for (i = 0; i < channels; i++) { + for (j = 0; j < subchans; j++) { + u32 *dst = get_dma_ptr_x(runtime, shift, i, j, hwoff); -+ u32 *src = (u32 *)buf + j * channels + i; ++ u32 __user *src = (u32 __user *)buf + j * channels + i; + for (k = 0; k < frames; k++, dst++, src += voices) + unsafe_get_user(*dst, src, faulted); + } -- 2.40.0.152.g15d061e6df

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[PATCH 0/2] kselftest/alsa: Decrease pcm-test duration to avoid timeouts
by Nícolas F. R. A. Prado 13 Jul '23

13 Jul '23

This series decreases the pcm-test duration in order to avoid timeouts by first moving the audio stream duration to a variable and subsequently decreasing it. Nícolas F. R. A. Prado (2): kselftest/alsa: pcm-test: Move stream duration and margin to variables kselftest/alsa: pcm-test: Decrease stream duration from 4 to 2 seconds tools/testing/selftests/alsa/pcm-test.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) -- 2.41.0

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