December 2022 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH 2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible
by Mark Brown 27 Dec '22

27 Dec '22

On Wed, 14 Dec 2022 04:40:58 +0000, David Rau wrote: > This adds the DAI mono mode support and set the frame width to 32 > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [2/2] ASoC: da7213: Add support for mono, set frame width to 32 when possible commit: c89e652e84f636354213aae42490c7f77a0eadde All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[PATCH 0/4] Add support for DSP volume controls
by Amadeusz Sławiński 27 Dec '22

27 Dec '22

Some devices like DMIC don't expose native controls or need volume limit due to possible HW damage. Add support for volume controls allowing to change volume level in DSP. Maximum volume level is imposed by the topology file which defines given path. Amadeusz Sławiński (4): ASoC: Intel: avs: Add peakvol runtime-parameter requests ASoC: Intel: avs: Add control volume operations ASoC: Intel: avs: Parse control tuples ASoC: Intel: avs: Peakvol module configuration include/uapi/sound/intel/avs/tokens.h | 4 + sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/control.c | 105 ++++++++++++++++++++++++++ sound/soc/intel/avs/control.h | 23 ++++++ sound/soc/intel/avs/messages.c | 29 +++++++ sound/soc/intel/avs/messages.h | 33 ++++++++ sound/soc/intel/avs/path.c | 62 +++++++++++++++ sound/soc/intel/avs/topology.c | 76 +++++++++++++++++++ sound/soc/intel/avs/topology.h | 3 + 9 files changed, 336 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/avs/control.c create mode 100644 sound/soc/intel/avs/control.h -- 2.25.1

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[PATCH] ASoC: SOF: amd: Enable cache for AMD Rembrandt platform
by V sujith kumar Reddy 27 Dec '22

27 Dec '22

Enable DSP cache for ACP memory Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/sof/amd/acp-dsp-offset.h | 4 ++++ sound/soc/sof/amd/acp-loader.c | 7 +++++++ sound/soc/sof/amd/acp.h | 2 ++ 3 files changed, 13 insertions(+) diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index de5726251dc6..920155dee819 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -85,4 +85,8 @@ #define ACP_SCRATCH_REG_0 0x10000 #define ACP6X_DSP_FUSION_RUNSTALL 0x0644 + +/* Cache window registers */ +#define ACP_DSP0_CACHE_OFFSET0 0x0420 +#define ACP_DSP0_CACHE_SIZE0 0x0424 #endif diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index 090c8b18c83c..a4bce5a3ae48 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -151,6 +151,7 @@ static void configure_pte_for_fw_loading(int type, int num_pages, struct acp_dev int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); struct acp_dev_data *adata; unsigned int src_addr, size_fw; u32 page_count, dma_size; @@ -183,6 +184,12 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) if (ret < 0) dev_err(sdev->dev, "acp dma transfer status: %d\n", ret); + if (desc->rev > 3) { + /* Cache Window enable */ + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_OFFSET0, desc->sram_pte_offset); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_SIZE0, SRAM1_SIZE | BIT(31)); + } + /* Free memory once DMA is complete */ dma_size = (PAGE_ALIGN(sdev->basefw.fw->size) >> PAGE_SHIFT) * ACP_PAGE_SIZE; dma_free_coherent(&pci->dev, dma_size, adata->bin_buf, adata->sha_dma_addr); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 09e16ef8afa0..4314094a97fd 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -72,6 +72,8 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 #define AMD_STACK_DUMP_SIZE 32 +#define SRAM1_SIZE 0x13A000 + enum clock_source { ACP_CLOCK_96M = 0, ACP_CLOCK_48M, -- 2.25.1

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[PATCH 00/15] ASoC: Drop empty platform remove functions
by Uwe Kleine-König 27 Dec '22

27 Dec '22

From: Uwe Kleine-König <uwe(a)kleine-koenig.org> Hello, this patch series removes all platform remove functions that only return zero below sound/soc. There is no reason to have these, as the only caller is platform core code doing: if (drv->remove) { int ret = drv->remove(dev); ... } (in platform_remove()) and so having no remove function is both equivalent and simpler. Best regards Uwe Uwe Kleine-König (15): ASoC: pxa: Drop empty platform remove function ASoC: amd: Drop empty platform remove function ASoC: atmel-classd: Drop empty platform remove function ASoC: atmel-pdmic: Drop empty platform remove function ASoC: 88pm860x: Drop empty platform remove function ASoC: ac97: Drop empty platform remove function ASoC: adau7002: Drop empty platform remove function ASoC: bt-sco: Drop empty platform remove function ASoC: cq93vc: Drop empty platform remove function ASoC: mc13783: Drop empty platform remove function ASoC: wl1273: Drop empty platform remove function ASoC: pxa: e740_wm9705: Drop empty platform remove function ASoC: pxa: e750_wm9705: Drop empty platform remove function ASoC: pxa: e800_wm9712: Drop empty platform remove function ASoC: SOF: nocodec: Drop empty platform remove function sound/soc/amd/raven/acp3x-i2s.c | 8 -------- sound/soc/atmel/atmel-classd.c | 6 ------ sound/soc/atmel/atmel-pdmic.c | 6 ------ sound/soc/codecs/88pm860x-codec.c | 6 ------ sound/soc/codecs/ac97.c | 6 ------ sound/soc/codecs/adau7002.c | 6 ------ sound/soc/codecs/bt-sco.c | 6 ------ sound/soc/codecs/cq93vc.c | 6 ------ sound/soc/codecs/mc13783.c | 6 ------ sound/soc/codecs/wl1273.c | 6 ------ sound/soc/pxa/e740_wm9705.c | 6 ------ sound/soc/pxa/e750_wm9705.c | 6 ------ sound/soc/pxa/e800_wm9712.c | 6 ------ sound/soc/pxa/spitz.c | 6 ------ sound/soc/sof/nocodec.c | 6 ------ 15 files changed, 92 deletions(-) base-commit: 3a28c2c89f4b412b648761430720d40a8dc326ef -- 2.38.1

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[PATCH] ASoC: pxa: Drop empty platform remove function
by Uwe Kleine-König 27 Dec '22

27 Dec '22

A remove callback just returning 0 is equivalent to no remove callback at all. So drop the useless function. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/pxa/spitz.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 44303b6eb228..70442315f5c5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -306,18 +306,12 @@ static int spitz_probe(struct platform_device *pdev) return ret; } -static int spitz_remove(struct platform_device *pdev) -{ - return 0; -} - static struct platform_driver spitz_driver = { .driver = { .name = "spitz-audio", .pm = &snd_soc_pm_ops, }, .probe = spitz_probe, - .remove = spitz_remove, }; module_platform_driver(spitz_driver); -- 2.38.1

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[PATCH] ASoC: dt-bindings: adi,adau7002: Convert to DT schema
by Krzysztof Kozlowski 27 Dec '22

27 Dec '22

Convert the Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter bindings to DT schema. During the conversion, add properties already used by DTS (sc7180-trogdor-coachz.dts) and Linux driver: 1. wakeup-delay-ms, 2. sound-dai-cells (via referencing dai-common.yaml). Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../bindings/sound/adi,adau7002.txt | 19 --------- .../bindings/sound/adi,adau7002.yaml | 40 +++++++++++++++++++ 2 files changed, 40 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt deleted file mode 100644 index f144ee1abf85..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau7002.txt +++ /dev/null @@ -1,19 +0,0 @@ -Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter - -Required properties: - - - compatible: Must be "adi,adau7002" - -Optional properties: - - - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD - supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt - - If this property is not present it is assumed that the supply pin is - hardwired to always on. - -Example: - adau7002: pdm-to-i2s { - compatible = "adi,adau7002"; - IOVDD-supply = <&supply>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.yaml b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml new file mode 100644 index 000000000000..fcca0fde7d86 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau7002.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: adi,adau7002 + + IOVDD-supply: + description: + IOVDD power supply, if skipped then it is assumed that the supply pin is + hardwired to always on. + + wakeup-delay-ms: + description: + Delay after power up needed for device to settle. + +required: + - compatible + +unevaluatedProperties: false + +examples: + - | + audio-codec { + compatible = "adi,adau7002"; + IOVDD-supply = <&pp1800_l15a>; + #sound-dai-cells = <0>; + wakeup-delay-ms = <80>; + }; -- 2.34.1

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Re: [PATCH 1/2] ASoC: rt5640: Allow configuration of LOUT to mono differential mode
by Mark Brown 27 Dec '22

27 Dec '22

On Fri, 09 Dec 2022 10:56:20 +0000, Jarrah Gosbell wrote: > LOUT output can produce either single-ended stereo signals, or mono > differential signal. Some applications are wired to use LOUT in mono > differential mode. Allow to configure it via device property. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: rt5640: Allow configuration of LOUT to mono differential mode commit: 44b54f54b41d36e8a2302a2c18a65268751c9067 [2/2] dt-bindings: sound: rt5640: Allow to describe how LOUT is wired commit: d8510532ccf3bc12e2b1c0c285d4b61d4b713e1a All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[PATCH] ALSA: hda/realtek: Apply dual codec fixup for Dell Latitude laptops
by Chris Chiu 27 Dec '22

27 Dec '22

The Dell Latiture 3340/3440/3540 laptops with Realtek ALC3204 have dual codecs and need the ALC1220_FIXUP_GB_DUAL_CODECS to fix the conflicts of Master controls. The existing headset mic fixup for Dell is also required to enable the jack sense and the headset mic. Introduce a new fixup to fix the dual codec and headset mic issues for particular Dell laptops since other old Dell laptops with the same codec configuration are already well handled by the fixup in alc269_fallback_pin_fixup_tbl[]. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e443d88f627f..3794b522c222 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7175,6 +7175,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, + ALC236_FIXUP_DELL_DUAL_CODECS, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9130,6 +9131,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, }, + [ALC236_FIXUP_DELL_DUAL_CODECS] = { + .type = HDA_FIXUP_PINS, + .v.func = alc1220_fixup_gb_dual_codecs, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9232,6 +9239,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c1c, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c1d, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.25.1

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[PATCH v2 0/3] ASoC: SOF: Fixes for suspend after firmware crash
by Peter Ujfalusi 27 Dec '22

27 Dec '22

Hi, This is the followup series for the v1 sent out by Ranjani [1]. Unfortunately Ranjani was dragged away to another issue and could not send the update herself. Changes since v1: - In patch 2, move the tear_down_all_pipelines call instead of duplicating it Amadeusz: I have kept the check as it is: if (tplg_ops && tplg_ops->tear_down_all_pipelines) I'm preparing the ops optionality change series which would require this change. [1] https://lore.kernel.org/alsa-devel/20221215185347.1457541-1-ranjani.sridhar… Regards, Peter --- Curtis Malainey (1): ASoC: SOF: Add FW state to debugfs Ranjani Sridharan (2): ASoC: SOF: pm: Set target state earlier ASoC: SOF: pm: Always tear down pipelines before DSP suspend sound/soc/sof/debug.c | 4 +++- sound/soc/sof/pm.c | 9 ++++----- 2 files changed, 7 insertions(+), 6 deletions(-) -- 2.39.0

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[PATCH 0/3] ASoC: SOF: Fixes for suspend after firmware crash
by Ranjani Sridharan 27 Dec '22

27 Dec '22

This series contains 2 patches to fix device suspend after a firmware crash and another patch to allow reading the FW state from debugfs. Curtis Malainey (1): ASoC: SOF: Add FW state to debugfs Ranjani Sridharan (2): ASoC: SOF: pm: Set target state earlier ASoC: SOF: pm: Always tear down pipelines before DSP suspend sound/soc/sof/debug.c | 4 +++- sound/soc/sof/pm.c | 6 ++++-- 2 files changed, 7 insertions(+), 3 deletions(-) -- 2.25.1

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