August 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/9] ASoC: sun8i-codec driver cleanup
by Samuel Holland 07 Sep '20

07 Sep '20

Now that the fixes series is merged, here is a series of small cleanups to the sun8i-codec driver. These help shorten the patch stack for the next series, which will add support for the other two DAIs in this codec: AIF2 and AIF3. Samuel Holland (9): ASoC: sun8i-codec: Remove extraneous widgets ASoC: sun8i-codec: Fix AIF1 MODCLK widget name ASoC: sun8i-codec: Fix AIF1_ADCDAT_CTRL field names ASoC: sun8i-codec: Fix AIF1_MXR_SRC field names ASoC: sun8i-codec: Fix ADC_DIG_CTRL field name ASoC: sun8i-codec: Fix field bit number indentation ASoC: sun8i-codec: Sort masks in a consistent order ASoC: sun8i-codec: Attach the bus clock to the regmap ASoC: sun8i-codec: Manage module clock via DAPM sound/soc/sunxi/sun8i-codec.c | 104 ++++++++++++---------------------- 1 file changed, 37 insertions(+), 67 deletions(-) -- 2.26.2

3 12

[PATCH v5 0/2] Add mediatek codec mt6359 driver
by Jiaxin Yu 07 Sep '20

07 Sep '20

Add mediatek codec (MT6359) driver MT6359 support playback and capture feature. On downlink path, it includes three DACs for handset, headset, and lineout path. On unlink path, it includeds three ADCs for main mic, second mic, 3rd mic, and headset mic. By scenario, select *_MUX widget to create damp path. And by select mic_type_mux to choose DMIC/AMIC/.... For example, select these MUX widget to create headset path (1) DAC In Mux --> "Normal Path" (2) HP Mux --> "Audio Playback" v5 changes: 1. Don't need to unregister the component which is already relegated to devm. v4 changes: 1. Add a remove() function to undo regulator_enable(). 2. Removed unnecessary logs. 3. patchwork link: https://patchwork.kernel.org/cover/11715553/ https://patchwork.kernel.org/patch/11716015/ https://patchwork.kernel.org/patch/11715557/ v3 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11714291/ https://patchwork.kernel.org/patch/11714295/ https://patchwork.kernel.org/patch/11714339/ v2 changes: 1. patchwork link: https://patchwork.kernel.org/cover/11706935/ https://patchwork.kernel.org/patch/11708865/ https://patchwork.kernel.org/patch/11706937/ v1 changes: 1.lkml link: https://lkml.org/lkml/2020/3/5/1257 Jiaxin Yu (2): ASoC: mediatek: mt6359: add codec driver dt-bindings: mediatek: mt6359: add codec document .../devicetree/bindings/sound/mt6359.yaml | 68 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6359.c | 2753 ++++++++++++++++++++ sound/soc/codecs/mt6359.h | 2640 +++++++++++++++++++ 5 files changed, 5471 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt6359.yaml create mode 100644 sound/soc/codecs/mt6359.c create mode 100644 sound/soc/codecs/mt6359.h -- 1.8.1.1.dirty

3 10

[PATCH 00/16] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This a merge of serie: - Add Add H6 I2S support https://patchwork.kernel.org/cover/11497007/ - Add Allwinner H3/H5/A64 HDMI audio https://patchwork.kernel.org/cover/11510511/ I merge both serie because there is a similar issue regarding the I2S polarity. This need to be investigated under a scope and see if the comment is still true for Allwinner H6 and previous SoC. LibreElec team found that we have to introduce the "simple-audio-card,frame-inversion" property with recent mainline change. In order to make HDMI audio work. Maybe the I2S polarity is good or maybe the silicium has an issue but not present for HDMI I2S or not present in TDM mode... I cannot do it myself, so if someone want to to do it please feel free. Regarding the discussion we had here: https://patchwork.kernel.org/patch/11510521/ I didn't switch to generic hdmi card-name and used name like: sun8i-h3-hdmi, sun50i-h6-hdmi, etc.... Despite this wrong comment the rest of the serie introduce some fix that should be merged even if it's without the H6 support. Regards Clement Change since v1: - drop allwinner,playback-channels property - use coherent hdmi,card-name - indentation fix - collect tags Clément Péron (2): ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (11): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 33 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 35 +++ .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 33 ++ sound/soc/sunxi/sun4i-i2s.c | 281 +++++++++++++++++- 9 files changed, 410 insertions(+), 6 deletions(-) -- 2.25.1

6 39

[PATCH v3 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 04 Sep '20

04 Sep '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: Add support for audio over DP ASoC: qcom: Optimise lpass variant structure .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 +- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 25 +- sound/soc/qcom/lpass-cpu.c | 92 ++- sound/soc/qcom/lpass-hdmi.c | 685 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 ++++ sound/soc/qcom/lpass-ipq806x.c | 25 +- sound/soc/qcom/lpass-lpaif-reg.h | 51 +- sound/soc/qcom/lpass-platform.c | 287 +++++++-- sound/soc/qcom/lpass-sc7180.c | 147 ++++- sound/soc/qcom/lpass.h | 123 +++- 13 files changed, 1472 insertions(+), 151 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 13

[PATCH v3 0/7] Add H6 I2S support
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v2 (thanks Maxime): - Add details in commit log about sign extend sample - Only set FIFO regs as volatile in regmap - Missing a space (detected by checkpatch) Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 265 +++++++++++++++++- 3 files changed, 296 insertions(+), 2 deletions(-) -- 2.20.1

6 28

[PATCH 1/2] dt-bindings: sound: midas-audio: Correct parsing sound-dai phandles
by Krzysztof Kozlowski 03 Sep '20

03 Sep '20

The "sound-dai" property has cells therefore phandle-array should be used, even if it is just one phandle. This fixes dtbs_check warnings like: arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0:1: missing phandle tag in 0 arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0: [158, 0] is too long Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../devicetree/bindings/sound/samsung,midas-audio.yaml | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 1c755de686f7..578928e67e5c 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -21,7 +21,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the I2S controller required: - sound-dai @@ -30,7 +31,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the WM1811 CODEC required: - sound-dai -- 2.17.1

4 13

[PATCH v4 0/3] ASoC: soundwire: fix port_ready[] dynamic allocation
by Bard Liao 03 Sep '20

03 Sep '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v4: - Reorder the patches order to allow bisect. Changes in v3: - Add ASoC tag in the cover letter title. - Edit the title and commit message of the third patch for better understanding. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): ASoC: codecs: soundwire: remove port_ready[] usage from codecs. soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

3 5

a saner API for allocating DMA addressable pages
by Christoph Hellwig 03 Sep '20

03 Sep '20

Hi all, this series replaced the DMA_ATTR_NON_CONSISTENT flag to dma_alloc_attrs with a separate new dma_alloc_pages API, which is available on all platforms. In addition to cleaning up the convoluted code path, this ensures that other drivers that have asked for better support for non-coherent DMA to pages with incurring bounce buffering over can finally be properly supported. I'm still a little unsure about the API naming, as alloc_pages sort of implies a struct page return value, but we return a kernel virtual address. The other alternative would be to name the API dma_alloc_noncoherent, but the whole non-coherent naming seems to put people off. As a follow up I plan to move the implementation of the DMA_ATTR_NO_KERNEL_MAPPING flag over to this framework as well, given that is also is a fundamentally non coherent allocation. The replacement for that flag would then return a struct page, as it is allowed to actually return pages without a kernel mapping as the name suggested (although most of the time they will actually have a kernel mapping..) In addition to the conversions of the existing non-coherent DMA users the last three patches also convert the DMA coherent allocations in the NVMe driver to use this new framework through a dmapool addition. This was both to give me a good testing vehicle, but also because it should speed up the NVMe driver on platforms with non-coherent DMA nicely, without a downside on platforms with cache coherent DMA. A git tree is available here: git://git.infradead.org/users/hch/misc.git dma_alloc_pages Gitweb: http://git.infradead.org/users/hch/misc.git/shortlog/refs/heads/dma_alloc_p… Diffstat: Documentation/core-api/dma-api.rst | 92 ++---- Documentation/core-api/dma-attributes.rst | 8 Documentation/userspace-api/media/v4l/buffer.rst | 17 - Documentation/userspace-api/media/v4l/vidioc-reqbufs.rst | 1 arch/alpha/kernel/pci_iommu.c | 2 arch/arm/include/asm/dma-direct.h | 4 arch/arm/mm/dma-mapping-nommu.c | 2 arch/arm/mm/dma-mapping.c | 4 arch/ia64/Kconfig | 3 arch/ia64/hp/common/sba_iommu.c | 2 arch/ia64/kernel/dma-mapping.c | 14 arch/ia64/mm/init.c | 3 arch/mips/Kconfig | 1 arch/mips/bmips/dma.c | 4 arch/mips/cavium-octeon/dma-octeon.c | 4 arch/mips/include/asm/dma-direct.h | 4 arch/mips/include/asm/jazzdma.h | 2 arch/mips/jazz/jazzdma.c | 102 +------ arch/mips/loongson2ef/fuloong-2e/dma.c | 4 arch/mips/loongson2ef/lemote-2f/dma.c | 4 arch/mips/loongson64/dma.c | 4 arch/mips/mm/dma-noncoherent.c | 48 +-- arch/mips/pci/pci-ar2315.c | 4 arch/mips/pci/pci-xtalk-bridge.c | 4 arch/mips/sgi-ip32/ip32-dma.c | 4 arch/parisc/Kconfig | 1 arch/parisc/kernel/pci-dma.c | 6 arch/powerpc/include/asm/dma-direct.h | 4 arch/powerpc/kernel/dma-iommu.c | 2 arch/powerpc/platforms/ps3/system-bus.c | 4 arch/powerpc/platforms/pseries/vio.c | 2 arch/s390/pci/pci_dma.c | 2 arch/x86/kernel/amd_gart_64.c | 8 drivers/gpu/drm/exynos/exynos_drm_gem.c | 2 drivers/gpu/drm/nouveau/nvkm/subdev/instmem/gk20a.c | 3 drivers/iommu/dma-iommu.c | 2 drivers/iommu/intel/iommu.c | 6 drivers/media/common/videobuf2/videobuf2-core.c | 36 -- drivers/media/common/videobuf2/videobuf2-dma-contig.c | 19 - drivers/media/common/videobuf2/videobuf2-dma-sg.c | 3 drivers/media/common/videobuf2/videobuf2-v4l2.c | 12 drivers/net/ethernet/amd/au1000_eth.c | 15 - drivers/net/ethernet/i825xx/lasi_82596.c | 36 +- drivers/net/ethernet/i825xx/lib82596.c | 148 +++++----- drivers/net/ethernet/i825xx/sni_82596.c | 23 - drivers/net/ethernet/seeq/sgiseeq.c | 24 - drivers/nvme/host/pci.c | 79 ++--- drivers/parisc/ccio-dma.c | 2 drivers/parisc/sba_iommu.c | 2 drivers/scsi/53c700.c | 120 ++++---- drivers/scsi/53c700.h | 9 drivers/scsi/sgiwd93.c | 14 drivers/xen/swiotlb-xen.c | 2 include/linux/dma-direct.h | 55 ++- include/linux/dma-mapping.h | 32 +- include/linux/dma-noncoherent.h | 21 - include/linux/dmapool.h | 23 + include/linux/gfp.h | 6 include/media/videobuf2-core.h | 3 include/uapi/linux/videodev2.h | 2 kernel/dma/Kconfig | 9 kernel/dma/Makefile | 1 kernel/dma/coherent.c | 17 + kernel/dma/direct.c | 112 +++++-- kernel/dma/mapping.c | 104 ++----- kernel/dma/ops_helpers.c | 86 ++++++ kernel/dma/pool.c | 2 kernel/dma/swiotlb.c | 4 kernel/dma/virt.c | 2 mm/dmapool.c | 211 +++++++++------ sound/mips/hal2.c | 58 +--- 71 files changed, 872 insertions(+), 803 deletions(-)

8 76

[PATCH v4 00/10] topology: decode: Various fixes
by Piotr Maziarz 03 Sep '20

03 Sep '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability v3: -No functional changes -Changed UCM to more descriptive standard ALSA configuration files -removed Gerrit's Change-Id -Added missing signed-offs v4: -Resend, added Reviewed-by tags Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

2 13

[PATCH] ALSA: hda/realtek - Add control fixup for Lenovo Thinkpad X1 Carbon 7th
by Benjamin Poirier 03 Sep '20

03 Sep '20

As a result of commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"), the sound output level on my machine, an X1 Carbon 7th gen, was reduced to ~65% of its previous level when playing certain sounds. [1] Internally, this laptop model has three outputs (PCM-OUT1, connection 0x02; PCM-OUT2, connection 0x03; SP-OUT PCM, connection 0x06) which can be routed to two sets of stereo speakers: Front (tweeters, node 0x14) and Bass (woofers, node 0x17, aka Rear in some contexts) and one headphone output (node 0x21). The tweeters are noticeably less powerful than the woofers. [2] Before commit d2cd795c4ece, the bass speakers were connected to SP-OUT PCM. SP-OUT PCM is meant for s/pdif output and does not have volume control. This connection made volume control commonly ineffective (using the Master slider in alsa or pulseaudio apparently had little effect or alternated between mute or max with nothing in between). commit d2cd795c4ece added quirk ALC285_FIXUP_SPEAKER2_TO_DAC1 which resulted in assigning both sets of speakers to PCM-OUT1, bringing the two sets of speakers under one effective volume control but also lowering the output volume noticeably. Fix this by connecting PCM-OUT1 to Front speakers and PCM-OUT2 to Rear speakers. The max output volume is restored to what it was before commit d2cd795c4ece. This is done by setting the connection of node 0x17 to 0x03. However, when we do this, the HDA auto config automatically changes the connection of node 0x21 to 0x02. This output, meant for the front speakers, has some "secret" equalizer which changes the output volume according to the level of what's being played, after some delay[3]. This is undesirable with headphones. Therefore, this patch manually limits the connection of node 0x21 to 0x03. Even though there are three speaker/line outputs (nodes 0x14, 0x17, 0x21), there are only two amps for volume control (nodes 0x02, 0x03). When setting node 0x17 to connection 0x03 and node 0x21 to connection 0x03, amp 0x03 is shared by the Woofer and Headphone outputs. Therefore, they cannot have independent volume control. This patch uses the Speaker control element to regulate nodes 0x02 (acts on tweeters) and 0x03 (acts on woofers and headphones). A virtual control element is created for the Headphone mixer which mirrors the Speaker mixer. These changes are only for volume controls; there are 3 hardware mute controls. The current fixup chain applied to this machine is ALC285_FIXUP_THINKPAD_HEADSET_JACK ALC285_FIXUP_SPEAKER2_TO_DAC1 ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE The fixup added by this patch is an alternative to ALC285_FIXUP_SPEAKER2_TO_DAC1 which must be selected explicitly using the module parameter snd_sof_intel_hda_common.hda_model=alc285-tpx1-dual-speakers when using the new SOF driver architecture, or snd_hda_intel.model=alc285-tpx1-dual-speakers when using the snd_hda_intel driver in legacy mode (see the snd_intel_dspcfg.dsp_driver module parameter). In order to be able to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK without changing current behavior, a new variant of that fixup is introduced. When using the model added by this patch, the fixup chain is: ALC285_FIXUP_TPX1_DUAL_SPEAKERS ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… [2] https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 [3] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Tested-by: Even Brenden <evenbrenden(a)gmail.com> Tested-by: Vincent Bernat <vincent(a)bernat.ch> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- This is an updated version of the patchset (now a single patch) to fix the low output volume on the Lenovo ThinkPad X1 Carbon 7th gen. With this patch, the volume output with the Linux driver matches the one with the Windows driver. Volume control is effective and headphone output is unaffected. Multiple users of this machine have reported a low output volume [1] when using a kernel with commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Many have tried these patches or a similar workaround using `hda-verb` and prefer this configuration[2]. [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... [2] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... Changes: v1 == https://lore.kernel.org/alsa-devel/20200211055651.4405-1-benjamin.poirier@g… v2 == https://lore.kernel.org/alsa-devel/20200703080005.8942-1-benjamin.poirier@g… * patch 1 Shortened log * patch 2 Change log to replace experimental observations by information from diagram posted on kernel.org bugzilla. Select exactly the desired connection for node 0x17 (bass speakers) instead of reusing alc295_fixup_disable_dac3(). Control node 0x21 (headphones) connection to avoid the volume "wobble" effect described in the log. Include SET_CONNECT_SEL command in fixup since it is not done by the hda core. Rename one control to reflect its new function and to use as a condition in the ucm config. v3 == * Remove patch 1 which was merged separately as commit 9774dc218bb6 ("ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id") * Make fixup activation manual using "alc285-tpx1-dual-speakers" model instead of automatic based on codec id. Suggested by Jaroslav. * Control both tweeter and woofer volume from "Speaker Playback Volume" control. This way a separate ucm change is not needed, unlike in v2 of the patchset. Suggested by Jaroslav. sound/pci/hda/patch_realtek.c | 355 +++++++++++++++++++++++++++++++++- 1 file changed, 351 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..8012c4aaf282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5821,6 +5821,338 @@ static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, } } +static struct snd_kcontrol *find_ctl_verbose(struct hda_codec *codec, + const char *name) +{ + struct snd_kcontrol *kctl; + + kctl = snd_hda_find_mixer_ctl(codec, name); + if (!kctl) + codec_warn(codec, "Did not find control \"%s\"\n", name); + return kctl; +} + +struct tpx1_dual_speaker { + struct hda_codec *codec; + struct snd_kcontrol underlying, hp_vol; +}; + +static int tpx1_dual_speaker_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.info(&speaker_priv->underlying, uinfo); +} + +static int tpx1_dual_speaker_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.get(&speaker_priv->underlying, + ucontrol); +} + +static int tpx1_dual_speaker_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + int err; + + /* Control tweeter volume */ + err = speaker_priv->underlying.put(&speaker_priv->underlying, + ucontrol); + if (err < 0) + return err; + + /* Control woofer volume (shared with headphone) */ + err = speaker_priv->hp_vol.put(&speaker_priv->hp_vol, ucontrol); + if (err < 0) + return err; + + snd_ctl_notify(speaker_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &speaker_priv->hp_vol.id); + return err; +} + +static int tpx1_dual_speaker_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.tlv.c(&speaker_priv->underlying, + op_flag, size, tlv); +} + +static void tpx1_dual_speaker_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + if (speaker_priv->underlying.private_free) + speaker_priv->underlying.private_free( + &speaker_priv->underlying); + kfree(speaker_priv); +} + +static int tpx1_dual_override_speaker_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_speaker *speaker_priv; + + speaker_priv = kmalloc(sizeof(struct tpx1_dual_speaker), GFP_KERNEL); + if (!speaker_priv) + return -ENOMEM; + speaker_priv->codec = codec; + memcpy(&speaker_priv->underlying, speaker_vol, + sizeof(struct snd_kcontrol)); + memcpy(&speaker_priv->hp_vol, hp_vol, sizeof(struct snd_kcontrol)); + + speaker_vol->info = &tpx1_dual_speaker_vol_info; + speaker_vol->get = &tpx1_dual_speaker_vol_get; + speaker_vol->put = &tpx1_dual_speaker_vol_put; + speaker_vol->tlv.c = &tpx1_dual_speaker_vol_tlv; + speaker_vol->private_data = speaker_priv; + speaker_vol->private_free = &tpx1_dual_speaker_vol_free; + + return 0; +} + +struct tpx1_dual_hp { + struct hda_codec *codec; + struct snd_kcontrol underlying, *speaker_vol; +}; + +static int tpx1_dual_hp_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.info(&hp_priv->underlying, uinfo); +} + +static int tpx1_dual_hp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->speaker_vol->get(hp_priv->speaker_vol, ucontrol); +} + +static int tpx1_dual_hp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + int err; + + err = hp_priv->speaker_vol->put(hp_priv->speaker_vol, ucontrol); + if (err < 0) + return err; + snd_ctl_notify(hp_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &hp_priv->speaker_vol->id); + + return 0; +} + +static int tpx1_dual_hp_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.tlv.c(&hp_priv->underlying, op_flag, size, + tlv); +} + +static void tpx1_dual_hp_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + if (hp_priv->underlying.private_free) + hp_priv->underlying.private_free(&hp_priv->underlying); + kfree(hp_priv); +} + +static int tpx1_dual_override_hp_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_hp *hp_priv; + + hp_priv = kmalloc(sizeof(struct tpx1_dual_hp), GFP_KERNEL); + if (!hp_priv) + return -ENOMEM; + hp_priv->codec = codec; + memcpy(&hp_priv->underlying, hp_vol, sizeof(struct snd_kcontrol)); + hp_priv->speaker_vol = speaker_vol; + + hp_vol->info = &tpx1_dual_hp_vol_info; + hp_vol->get = &tpx1_dual_hp_vol_get; + hp_vol->put = &tpx1_dual_hp_vol_put; + hp_vol->tlv.c = &tpx1_dual_hp_vol_tlv; + hp_vol->private_data = hp_priv; + hp_vol->private_free = &tpx1_dual_hp_vol_free; + + return 0; +} + +/* We cannot use snd_hda_add_vmaster() because it expects the controls to be + * HDA (see init_follower_0dB()). This is not the case here. + */ +static int tpx1_dual_create_main_volume_ctl(struct hda_codec *codec) +{ + struct snd_kcontrol *main_vol, *speaker_vol; + struct snd_ctl_elem_value *ucontrol = NULL; + struct hda_gen_spec *spec = codec->spec; + struct snd_ctl_elem_info *uinfo; + int ch, err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return -ENOENT; + + main_vol = snd_ctl_make_virtual_master("Master Playback Volume", + spec->vmaster_tlv); + if (!main_vol) + return -ENOMEM; + err = snd_hda_ctl_add(codec, 0, main_vol); + if (err < 0) + return err; + err = snd_ctl_add_follower(main_vol, speaker_vol); + if (err < 0) + return err; + + /* Initialize values */ + uinfo = kmalloc(sizeof(struct snd_ctl_elem_info), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + + uinfo->id = speaker_vol->id; + err = speaker_vol->info(speaker_vol, uinfo); + if (err) + goto out; + + ucontrol = kmalloc(sizeof(struct snd_ctl_elem_value), GFP_KERNEL); + if (!ucontrol) { + err = -ENOMEM; + goto out; + } + + ucontrol->id = speaker_vol->id; + for (ch = 0; ch < uinfo->count; ch++) { + ucontrol->value.integer.value[ch] = + uinfo->value.integer.max; + } + err = speaker_vol->put(speaker_vol, ucontrol); + if (err < 0) + goto out; + else + err = 0; + +out: + kfree(ucontrol); + kfree(uinfo); + return err; +} + +static void alc285_fixup_tpx1_dual_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conn[] = { 0x03 }; + + /* For NID 0x17 (bass speakers), the connection list is {0x02, + * 0x03, 0x06}. Disable SP-OUT PCM (0x06) selection since it + * has no volume control, disable PCM1 (0x02) selection since + * it is for front speakers. This leaves PCM2 (0x03). + */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), + conn); + /* For NID 0x21 (headphone out), the connection list is {0x02, + * 0x03}. Disable LOUT1 (0x02) selection since its volume + * fluctuates according to input level. This leaves LOUT2 + * (0x03). + */ + snd_hda_override_conn_list(codec, 0x21, ARRAY_SIZE(conn), + conn); + + /* "Master" controls will be created manually after + * subordinate controls have been changed in the + * HDA_FIXUP_ACT_BUILD phase. + */ + spec->gen.suppress_vmaster = 1; + } else if (action == HDA_FIXUP_ACT_INIT) { + /* Because the overridden connection lists contain only a + * single node, __parse_nid_path() does not label the output + * as "multi". This leads snd_hda_activate_path() to skip the + * AC_VERB_SET_CONNECT_SEL even though it might be needed. Do + * it here instead. + * Note that when doing AC_VERB_SET_CONNECT_SEL, the + * connection is specified by index instead of nid. + */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + } else if (action == HDA_FIXUP_ACT_BUILD) { + struct snd_kcontrol *speaker_vol, *hp_vol; + int err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return; + + hp_vol = find_ctl_verbose(codec, "Headphone Playback Volume"); + if (!hp_vol) + return; + + err = tpx1_dual_override_speaker_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override speaker volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_override_hp_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override headphone volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_create_main_volume_ctl(codec); + if (err) { + codec_warn(codec, "Failed to create main volume control, err %d\n", + err); + return; + } + + err = __snd_hda_add_vmaster( + codec, "Master Playback Switch", NULL, + ((const char * const[]){ + "Speaker", "Bass Speaker", "Headphone", NULL + }), "Playback Switch", true, + &spec->gen.vmaster_mute.sw_kctl); + if (err) { + codec_warn(codec, "Failed to create main mute control, err %d\n", + err); + return; + } + + if (spec->gen.vmaster_mute.hook) { + snd_hda_add_vmaster_hook(codec, + &spec->gen.vmaster_mute, + spec->gen.vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->gen.vmaster_mute); + } + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -6083,6 +6415,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_TPX1_DUAL_SPEAKERS, ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, @@ -6135,7 +6468,8 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, @@ -6905,6 +7239,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_TPX1_DUAL_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_tpx1_dual_speakers, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI + }, [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, @@ -7280,12 +7620,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7741,8 +8087,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7934,6 +8280,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_TPX1_DUAL_SPEAKERS, .name = "alc285-tpx1-dual-speakers"}, {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, -- 2.28.0

3 6