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

[PATCH v2] ALSA: seq: Fix memory leak at error path in snd_seq_create_port()
by Takashi Iwai 17 Jul '23

17 Jul '23

We forgot to release a newly allocated item at the error path in snd_seq_create_port(). This patch fixes it. Fixes: 7c3f0d3d3a11 ("ALSA: seq: Check the conflicting port at port creation") Reported-by: syzbot+cf8e7fa4eeec59b3d485(a)syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000098ed3a0600965f89@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: only patch meta data correction sound/core/seq/seq_ports.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 9b80f8275026..f3f14ff0f80f 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -149,6 +149,7 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { + kfree(new_port); num = -EBUSY; goto unlock; } -- 2.35.3

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[GIT PULL] ASoC fixes for v6.5-rc1
by Mark Brown 17 Jul '23

17 Jul '23

The following changes since commit 2d0cad0473bd1ffbc5842be0b9f2546265acb011: ASoC: core: Always store of_node when getting DAI link component (2023-06-23 22:09:00 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc1 for you to fetch changes up to e8bf1741c14eb8e4a4e1364d45aeeab66660ab9b: MAINTAINERS: Redo addition of ssm3515 to APPLE SOUND (2023-07-15 13:07:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A lot of fixes here for the Qualcomm CODEC drivers, there was quite a bit of fragility with the SoundWire probe due to the combined DT and hotplug approach that the bus has which Johan Hovold fixed along with a bunch of other issues that came up in the process. Srivinvas Kandagatla also fixed some separate issues that have been lurking for a while in the Qualcomm AP side, and there's a good set of AMD fixes from Vijendar Mukunda too. ---------------------------------------------------------------- Derek Fang (1): ASoC: rt5640: Fix the issue of speaker noise Fabio Estevam (1): ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode" Geert Uytterhoeven (1): ASoC: codecs: SND_SOC_WCD934X should select REGMAP_IRQ Johan Hovold (15): ASoC: codecs: wcd938x: fix codec initialisation race ASoC: codecs: wcd938x: fix mbhc impedance loglevel ASoC: codecs: wcd938x: drop inline keywords ASoC: codecs: wcd938x: use dev_printk() for impedance logging ASoC: codecs: wcd934x: demote impedance printk ASoC: codecs: wcd934x: drop inline keywords ASoC: codecs: wcd938x: fix soundwire initialisation race ASoC: codecs: wcd938x: fix missing mbhc init error handling ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors Mario Limonciello (1): ASoC: amd: ps: Fix extraneous error messages Mark Brown (2): ASoC: codecs: wcd938x/wcd934x: loglevel fix and ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral Martin Povišer (1): MAINTAINERS: Redo addition of ssm3515 to APPLE SOUND Mastan Katragadda (1): ASoC: SOF: amd: add revision check for sending sha dma completion command Matthias Reichl (1): ASoC: hdmi-codec: fix channel info for compressed formats Nathan Chancellor (1): ASoC: cs35l45: Select REGMAP_IRQ Rob Herring (1): ASoC: dt-bindings: audio-graph-card2: Drop incomplete example Rohit kumar (1): ASoC: dt-bindings: Update maintainer email id Sameer Pujar (1): ASoC: rt5640: Fix sleep in atomic context Sheetal (2): ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map Shuming Fan (1): ASoC: rt5645: check return value after reading device id Srinivas Kandagatla (4): ASoC: qdsp6: q6apm: use dai link pcm id as pcm device number ASoC: qcom: q6afe-dai: fix Display Port Playback stream name ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR ASoC: qcom: q6apm: do not close GPR port before closing graph Vijendar Mukunda (6): ASoC: amd: ps: add comments for DMA irq bits mapping ASoC: amd: ps: add fix for dma irq mask for rx streams for SDW0 instance ASoC: amd: ps: fix for position register set for AUDIO0 RX stream ASoC: amd: ps: add comments for DMA register mapping ASoC: amd: ps: fix byte count return value for invalid SoundWire manager instance ASoC: amd: acp: fix for invalid dai id handling in acp_get_byte_count() .../bindings/sound/audio-graph-card2.yaml | 20 +---- .../bindings/sound/google,sc7180-trogdor.yaml | 2 +- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- MAINTAINERS | 2 + sound/soc/amd/acp/amd.h | 7 +- sound/soc/amd/ps/acp63.h | 22 ++++- sound/soc/amd/ps/pci-ps.c | 4 +- sound/soc/amd/ps/ps-sdw-dma.c | 16 +++- sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/hdmi-codec.c | 36 +++++--- sound/soc/codecs/rt5640.c | 13 +-- sound/soc/codecs/rt5645.c | 6 +- sound/soc/codecs/wcd-mbhc-v2.c | 57 +++++++++---- sound/soc/codecs/wcd934x.c | 20 ++++- sound/soc/codecs/wcd938x.c | 99 ++++++++++++++++++---- sound/soc/fsl/fsl_sai.c | 6 -- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 1 + sound/soc/qcom/qdsp6/q6apm.c | 7 +- sound/soc/qcom/qdsp6/topology.c | 4 +- sound/soc/soc-core.c | 6 +- sound/soc/soc-topology.c | 10 ++- sound/soc/sof/amd/acp.c | 10 ++- sound/soc/tegra/tegra210_adx.c | 34 +++++--- sound/soc/tegra/tegra210_amx.c | 40 +++++---- 25 files changed, 294 insertions(+), 134 deletions(-)

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[PATCH] ALSA: seq: Fix memory leak at error path in snd_seq_create_port()
by Takashi Iwai 17 Jul '23

17 Jul '23

We forgot to release a newly allocated item at the error path in snd_seq_create_port(). This patch fixes it. Reported-by: syzbot+cf8e7fa4eeec59b3d485(a)syzkaller.appspotmail.com Closes: https://lore.kernel.org/r/00000000000098ed3a0600965f89@google.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_ports.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 9b80f8275026..f3f14ff0f80f 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -149,6 +149,7 @@ int snd_seq_create_port(struct snd_seq_client *client, int port, write_lock_irq(&client->ports_lock); list_for_each_entry(p, &client->ports_list_head, list) { if (p->addr.port == port) { + kfree(new_port); num = -EBUSY; goto unlock; } -- 2.35.3

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Pass through encoded audio in the AC3 profile
by GitHub issues - opened 16 Jul '23

16 Jul '23

alsa-project/alsa-plugins issue #53 was opened from AndrewAmmerlaan: Is there any technical reason why it is not possible to `pactl set-sink-formats X "pcm dts-iec61937 ac3-iec61937"` when the `Digital Surround 5.1 IEC958/AC3 Output` profile is selected? It would be convenient to not have to switch profiles when watching a movie with audio that is already encoded AC3. I noticed the `dcaenc` plug-in for DTS encoding does the same thing. So maybe there is a good reason for this I'm not seeing. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/53 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH] ASoC: dt-bindings: audio-graph-card2: Drop incomplete example
by Rob Herring 16 Jul '23

16 Jul '23

The example in audio-graph-card2 binding is incomplete, uses undocumented compatibles strings, and doesn't follow typical .dts formatting. Rather than try to fix with what would probably be a lengthy example, just drop the example. Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/audio-graph-card2.yaml | 20 +------------------ 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml index 3de7b36829da..d3ce4de449d5 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-card2.yaml @@ -39,22 +39,4 @@ required: additionalProperties: false -examples: - - | - sound { - compatible = "audio-graph-card2"; - - links = <&cpu_port>; - }; - - cpu { - compatible = "cpu-driver"; - - cpu_port: port { cpu_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; - }; - - codec { - compatible = "codec-driver"; - - port { codec_ep: endpoint { remote-endpoint = <&cpu_ep>; }; }; - }; +... -- 2.40.1

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[PATCH -next] ASoC: amd: acp: clean up some inconsistent indentings
by Yang Li 16 Jul '23

16 Jul '23

sound/soc/amd/acp/acp-rembrandt.c:283 rmb_pcm_resume() warn: inconsistent indenting Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5863 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/amd/acp/acp-rembrandt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index 89314d95ec2b..21e67ed956d1 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -280,8 +280,8 @@ static int __maybe_unused rmb_pcm_resume(struct device *dev) } } } - spin_unlock(&adata->acp_lock); - return 0; + spin_unlock(&adata->acp_lock); + return 0; } static const struct dev_pm_ops rmb_dma_pm_ops = { -- 2.20.1.7.g153144c

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[PATCH] ALSA: Explicitly include correct DT includes
by Rob Herring 16 Jul '23

16 Jul '23

The DT of_device.h and of_platform.h date back to the separate of_platform_bus_type before it as merged into the regular platform bus. As part of that merge prepping Arm DT support 13 years ago, they "temporarily" include each other. They also include platform_device.h and of.h. As a result, there's a pretty much random mix of those include files used throughout the tree. In order to detangle these headers and replace the implicit includes with struct declarations, users need to explicitly include the correct includes. Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/atmel/ac97c.c | 3 +-- sound/drivers/serial-generic.c | 2 +- sound/pci/hda/hda_tegra.c | 3 ++- sound/ppc/awacs.c | 1 + sound/ppc/burgundy.c | 1 + sound/sparc/amd7930.c | 3 +-- sound/sparc/cs4231.c | 2 +- sound/sparc/dbri.c | 2 +- 8 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index c8912b8a1dc5..402b5f66dcc3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -12,13 +12,12 @@ #include <linux/gpio/consumer.h> #include <linux/init.h> #include <linux/interrupt.h> +#include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/mutex.h> #include <linux/types.h> #include <linux/io.h> -#include <linux/of.h> -#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> diff --git a/sound/drivers/serial-generic.c b/sound/drivers/serial-generic.c index e1f864dc7939..b0262541802a 100644 --- a/sound/drivers/serial-generic.c +++ b/sound/drivers/serial-generic.c @@ -16,7 +16,7 @@ #include <linux/io.h> #include <linux/ioport.h> #include <linux/module.h> -#include <linux/of_device.h> +#include <linux/of.h> #include <linux/serdev.h> #include <linux/serial_reg.h> #include <linux/slab.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 9d0ab043880b..39fa036616ce 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -16,7 +16,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/mutex.h> -#include <linux/of_device.h> +#include <linux/of.h> +#include <linux/platform_device.h> #include <linux/reset.h> #include <linux/slab.h> #include <linux/time.h> diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 53d558b2806c..659866cfe3b4 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -11,6 +11,7 @@ #include <asm/nvram.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of.h> #include <linux/slab.h> #include <sound/core.h> #include "pmac.h" diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 4fb990ab2ceb..400a886562b1 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -9,6 +9,7 @@ #include <linux/io.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of.h> #include <sound/core.h> #include "pmac.h" #include "burgundy.h" diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index c434b69a83f1..0fea04acc3ea 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -37,7 +37,7 @@ #include <linux/interrupt.h> #include <linux/moduleparam.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <linux/io.h> #include <sound/core.h> @@ -47,7 +47,6 @@ #include <sound/initval.h> #include <asm/irq.h> -#include <asm/prom.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 31bac355ec4d..c2ad3fa2f25a 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -18,7 +18,7 @@ #include <linux/irq.h> #include <linux/io.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 376aed136a45..050e98f32d36 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -69,7 +69,7 @@ #include <sound/initval.h> #include <linux/of.h> -#include <linux/of_device.h> +#include <linux/platform_device.h> #include <linux/atomic.h> #include <linux/module.h> -- 2.40.1

2 1

speaker-test: inconsistent behaviour of --buffer option.
by GitHub issues - edited 16 Jul '23

16 Jul '23

alsa-project/alsa-utils issue #224 was edited from AndrewAmmerlaan: Without specifying the `--buffer` option `speaker-test` returns this: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 0 - Front Left 1 - Front Right Write error: -77,File descriptor in bad state xrun_recovery failed: -77,File descriptor in bad state Transfer failed: File descriptor in bad state ``` Note that it says `Using max buffer size 1048576`. But now if we try to explicitly specify this maximum buffer size: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1048576 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251287 ``` Note that now we get `Requested buffer time 1000000 us` which is less then what we actually specified. In fact this will happen with any `--buffer` parameter larger then `1000000`: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1000001 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251042 ``` Leading me to believe that despite what `speaker-test` claims, `1000000` is actually the maximum, not `1048576`. Discovered while debugging https://gitlab.freedesktop.org/drm/intel/-/issues/8462 This behaviour of `speaker-test` is confusing though and feels to me like a separate issue. Issue URL : https://github.com/alsa-project/alsa-utils/issues/224 Repository URL: https://github.com/alsa-project/alsa-utils

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speaker-test: inconsistent behaviour of --buffer option.
by GitHub issues - opened 16 Jul '23

16 Jul '23

alsa-project/alsa-utils issue #224 was opened from AndrewAmmerlaan: Without specifying the `--buffer` option `speaker-test returns this: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Using max buffer size 1048576 Periods = 4 was set period_size = 262144 was set buffer_size = 1048576 0 - Front Left 1 - Front Right Write error: -77,File descriptor in bad state xrun_recovery failed: -77,File descriptor in bad state Transfer failed: File descriptor in bad state ``` Note that it says `Using max buffer size 1048576`. But now if we try to explicitly specify this maximum buffer size: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1048576 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251287 ``` Note that now we get `Requested buffer time 1000000 us` which is less then what we actually specified. In fact this will happen with any `--buffer` parameter larger then `1000000`: ``` andrew@andrew-gentoo-pc ~ % speaker-test -c2 -twav -Dplughw:CARD=PCH,DEV=3 --buffer 1000001 speaker-test 1.2.9 Playback device is plughw:CARD=PCH,DEV=3 Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 64 to 1048576 Period size range from 32 to 524288 Requested buffer time 1000000 us Periods = 4 was set period_size = 12000 was set buffer_size = 48000 0 - Front Left 1 - Front Right Time per period = 2.251042 ``` Leading me to believe that despite what `speaker-test` claims, `1000000` is actually the maximum, not `1048576`. Discovered while debugging https://gitlab.freedesktop.org/drm/intel/-/issues/8462 This behaviour of `speaker-test` is confusing though and feels to me like a separate issue. Issue URL : https://github.com/alsa-project/alsa-utils/issues/224 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH 1/2] ALSA: hda: cs35l41: Add fixups for machines without a valid ACPI _DSD
by David Xu 15 Jul '23

15 Jul '23

As the comments added in commit 4d4c4bff4f8ed79d95e05 ("ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties"), CSC3551 requires a valid _DSD to work and the current implementation just fails when no _DSD can be found for CSC3551. However it is a fact that many OEMs hardcoded the configurations needed by CSC3551 into their proprietary software for various 2022 and later laptop models, and this makes the Linux installations on these models cannot make any speaker sound. Meanwhile, at this point of time, we see no hope that these OEMs would ever fix this problem via a BIOS update. So to fix this bothering problem it might be worthwhile to add some model-specific fixups to apply some proper configurations to the cs35l41. To address the above problem, a configuration fixup function apply_cs35l41_fixup_cfg that would be called in cs35l41_no_acpi_dsd, along with a fixup table cs35l41_fixup_cfgtbl which is a array of fixup entry struct cs35l41_fixup_cfg are introduced. Each fixup entry records the ACPI _SUB(vender and device ID) to be matched, and a configuration to be applied to each of the cs35l41 device in CSC3551. More specifically for the design of this fixup mechanism, the maximum number of cs35l41 configurations inside a fixup entry is defined as a macro CS35L41_FIXUP_CFG_MAX_DEVICES, and the actual number of cs35l41 devices in a CSC3551 system is recorded in the num_device field in the fixup entry. The apply_cs35l41_fixup_cfg function is responsible for finding and applying a fixup for a specific model according to the model's ACPI _SUB. If no valid fixup can be applied, the fixup function fails and returns to the normal cs35l41_no_acpi_dsd execution flow. This patch now contains only several fixups for three Lenovo laptop models, namely 16IAH7, 16IAX7, and 16ARHA7 and this fixup mechanism has been verified to work on 16IAH7. As far as is known, several other laptop models from ASUS and HP also suffer from this no valid _DSD problem and could have it addressed with this fixup mechanism when proper fixup entries are inserted. Signed-off-by: David Xu <xuwd1(a)hotmail.com> --- sound/pci/hda/cs35l41_hda.c | 160 ++++++++++++++++++++++++++++++++++++ 1 file changed, 160 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index ce5faa620517..d957458dd4e6 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1211,6 +1211,159 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, return speaker_id; } +#define CS35L41_FIXUP_CFG_MAX_DEVICES 4 + +struct cs35l41_fixup_cfg { + unsigned short vender; + unsigned short device; + unsigned int num_device; /* The num of cs35l41 instances */ + /* cs35l41 instance ids, can be i2c index or spi index */ + int ids[CS35L41_FIXUP_CFG_MAX_DEVICES]; + unsigned int reset_gpio_idx[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum gpiod_flags reset_gpio_flags[CS35L41_FIXUP_CFG_MAX_DEVICES]; + int spkid_gpio_idx[CS35L41_FIXUP_CFG_MAX_DEVICES]; + unsigned int spk_pos[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_hda_gpio_function gpio1_func[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_hda_gpio_function gpio2_func[CS35L41_FIXUP_CFG_MAX_DEVICES]; + enum cs35l41_boost_type bst_type[CS35L41_FIXUP_CFG_MAX_DEVICES]; +}; + +static const struct cs35l41_fixup_cfg cs35l41_fixup_cfgtbl[] = { + { // Lenovo Legion Slim 7i 16IAH7 + .vender = 0x17aa, + .device = 0x386e, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_NOT_USED}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion Slim 7i 16IAH7 type2 + .vender = 0x17aa, + .device = 0x3803, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_NOT_USED}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion 7i 16IAX7 + .vender = 0x17aa, + .device = 0x3874, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion 7i 16IAX7 type 2 + .vender = 0x17aa, + .device = 0x386f, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + { // Lenovo Legion Slim 7 16ARHA7 + .vender = 0x17aa, + .device = 0x3877, + .num_device = 2, + .ids = {0x40, 0x41}, + .reset_gpio_idx = {0, 0}, + .reset_gpio_flags = {GPIOD_OUT_LOW, GPIOD_OUT_LOW}, + .spkid_gpio_idx = {1, 1}, + .spk_pos = {0, 1}, + .gpio1_func = {CS35l41_VSPK_SWITCH, CS35l41_VSPK_SWITCH}, + .gpio2_func = {CS35L41_INTERRUPT, CS35L41_INTERRUPT}, + .bst_type = {CS35L41_EXT_BOOST, CS35L41_EXT_BOOST} + }, + {} // terminator +}; + +static inline int cs35l41_fixup_get_index(const struct cs35l41_fixup_cfg *fixup, int cs35l41_addr) +{ + int i; + + for (i = 0; i < fixup->num_device; i++) { + if (fixup->ids[i] == cs35l41_addr) + return i; + } + return -ENODEV; +} + +static int apply_cs35l41_fixup_cfg(struct cs35l41_hda *cs35l41, + struct device *physdev, + int cs35l41_addr, + const struct cs35l41_fixup_cfg *fixup_tbl) +{ + const char *ssid; + unsigned int vendid; + unsigned int devid; + const struct cs35l41_fixup_cfg *cur_fixup; + struct cs35l41_hw_cfg *hw_cfg; + int cs35l41_index; + int ret; + int i; + + ssid = cs35l41->acpi_subsystem_id; + ret = sscanf(ssid, "%04x%04x", &vendid, &devid); + if (ret != 2) + return -EINVAL; + + hw_cfg = &cs35l41->hw_cfg; + for (cur_fixup = fixup_tbl; cur_fixup->vender; cur_fixup++) { + if (cur_fixup->vender == vendid && cur_fixup->device == devid) { + cs35l41_index = cs35l41_fixup_get_index(cur_fixup, cs35l41_addr); + if (cs35l41_index == -ENODEV) + return -ENODEV; + cs35l41->index = cs35l41_index; + cs35l41->reset_gpio = gpiod_get_index( + physdev, + NULL, + cur_fixup->reset_gpio_idx[cs35l41_index], + cur_fixup->reset_gpio_flags[cs35l41_index] + ); + cs35l41->speaker_id = cs35l41_get_speaker_id(physdev, + cs35l41_index, + cur_fixup->num_device, + cur_fixup->spkid_gpio_idx[cs35l41_index] + ); + hw_cfg->spk_pos = cur_fixup->spk_pos[cs35l41_index]; + cs35l41->channel_index = 0; + for (i = 0; i < cs35l41->index; i++) + if (cur_fixup->spk_pos[i] == hw_cfg->spk_pos) + cs35l41->channel_index++; + + hw_cfg->gpio1.func = cur_fixup->gpio1_func[cs35l41_index]; + hw_cfg->gpio1.valid = true; + hw_cfg->gpio2.func = cur_fixup->gpio2_func[cs35l41_index]; + hw_cfg->gpio2.valid = true; + hw_cfg->bst_type = cur_fixup->bst_type[cs35l41_index]; + dev_dbg(physdev, "Fixup applied.\n"); + break; + } + } + return 0; + +} + /* * Device CLSA010(0/1) doesn't have _DSD so a gpiod_get by the label reset won't work. * And devices created by serial-multi-instantiate don't have their device struct @@ -1221,6 +1374,7 @@ static int cs35l41_get_speaker_id(struct device *dev, int amp_index, static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physdev, int id, const char *hid) { + int ret; struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; /* check I2C address to assign the index */ @@ -1243,7 +1397,13 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd /* * Note: CLSA010(0/1) are special cases which use a slightly different design. * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + * However many OEMs hardcoded the configurations into their proprietary software + * thus leaving our Linux installation with no speaker sound at all while we see + * no hope those OEMs would fix it. So we apply a ssid specific fixup to fix it. */ + if (apply_cs35l41_fixup_cfg(cs35l41, physdev, id, cs35l41_fixup_cfgtbl) == 0) + return 0; + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; -- 2.41.0

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