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

[PATCH v3 0/3] Fix Tegra194 HDA regression
by Sameer Pujar 23 Dec '21

23 Dec '21

HDA probe failure is observed on Tegra194 based platforms and this happens due to reset failure. This series fixes the problem by skipping the failing reset and DT bindings are updated accordingly. Changelog ========= v2 -> v3: --------- * Use reset bulk APIs in HDA driver as suggested by Dmitry. v1 -> v2: --------- * Updated HDA driver patch to skip the failing reset instead of skipping resets in general for BPMP devices as per comment from Dmitry. * Used a better strucure name for SoC data as per comment from Thierry. * Dropped 'Fixes' tag in binding doc patch as per comment from Dmitry. Sameer Pujar (3): ALSA: hda/tegra: Fix Tegra194 HDA reset failure dt-bindings: sound: tegra: Update HDA resets arm64: tegra: Remove non existent Tegra194 reset .../bindings/sound/nvidia,tegra30-hda.yaml | 13 +++++-- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 5 +-- sound/pci/hda/hda_tegra.c | 45 +++++++++++++++++----- 3 files changed, 47 insertions(+), 16 deletions(-) -- 2.7.4

4 8

[GIT PULL] ASoC fixes for v5.16-rc6
by Mark Brown 22 Dec '21

22 Dec '21

The following changes since commit 3fc27e9a1f619b50700f020e6cd270c1b74755f0: ASoC: codecs: wsa881x: fix return values from kcontrol put (2021-12-01 14:13:53 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc6 for you to fetch changes up to ee907afb0c39a41ee74b862882cfe12820c74b98: ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s (2021-12-14 17:15:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 This is a relatively large set of driver specific changes so it may make sense to hold off to v5.17, though picking some over might be good. It's a combination of new device IDs and fixes for various driver specific things which are all small and of the usual "really bad if you're running into them" level, especially the Tegra ones. ---------------------------------------------------------------- Derek Fang (1): ASoC: rt5682: fix the wrong jack type detected Dmitry Osipenko (2): ASoC: tegra: Add DAPM switches for headphones and mic jack ASoC: tegra: Restore headphones jack name on Nyan Big Kai Vehmanen (2): ASoC: SOF: Intel: pci-tgl: add ADL-N support ASoC: SOF: Intel: pci-tgl: add new ADL-P variant Martin Blumenstingl (2): ASoC: meson: aiu: fifo: Add missing dma_coerce_mask_and_coherent() ASoC: meson: aiu: Move AIU_I2S_MISC hold setting to aiu-fifo-i2s Martin Povišer (1): ASoC: tas2770: Fix setting of high sample rates sound/soc/codecs/rt5682.c | 4 ++++ sound/soc/codecs/tas2770.c | 4 ++-- sound/soc/meson/aiu-encoder-i2s.c | 33 --------------------------------- sound/soc/meson/aiu-fifo-i2s.c | 19 +++++++++++++++++++ sound/soc/meson/aiu-fifo.c | 6 ++++++ sound/soc/sof/intel/pci-tgl.c | 4 ++++ sound/soc/tegra/tegra_asoc_machine.c | 11 ++++++++++- sound/soc/tegra/tegra_asoc_machine.h | 1 + 8 files changed, 46 insertions(+), 36 deletions(-)

2 1

[PATCH 0/2] ALSA: hda: intel-sdw-acpi: fix controller detection
by Pierre-Louis Bossart 22 Dec '21

22 Dec '21

The SoundWire-related information provided by platform firmware (DSDT) hasn't changed since 2016, but with a recent change of device hierarchy we need to change how the controller is detected. These patches were tested on existing devices and don't break anything, but are very much needed for newer devices (quirks already in Linus' branch for 5.16, see e.g. commit f55af7055cd4 ("ASoC: Intel: sof_sdw: Add support for SKU 0B12 product"). Unfortunately the BIOS changes happened *after* the initial patches were contributed. Libin Yang (2): ALSA: hda: intel-sdw-acpi: harden detection of controller ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2 sound/hda/intel-sdw-acpi.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) -- 2.25.1

2 3

[PATCH 0/2] ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2
by Bard Liao 22 Dec '21

22 Dec '21

In the HDAS ACPI scope, the SoundWire may not be the direct child of HDAS. It needs to go through the ACPI table at max depth of 2 to find the SoundWire device from HDAS. Libin Yang (2): ALSA: hda: intel-sdw-acpi: harden detection of controller ALSA: hda: intel-sdw-acpi: go through HDAS ACPI at max depth of 2 sound/hda/intel-sdw-acpi.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) -- 2.17.1

2 3

[PATCH] ALSA: rawmidi - fix the uninitalized user_pversion
by Jaroslav Kysela 22 Dec '21

22 Dec '21

The user_pversion was unitialized for the user space file structure in the open function, because the file private structure use kmalloc for the allocation. The kernel ALSA sequencer code clears the file structure, so no additional fixes are required. Cc: stable(a)kernel.org Cc: broonie(a)kernel.org BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Fixes: 09d23174402d ("ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION") Reported-by: syzbot+88412ee8811832b00dbe(a)syzkaller.appspotmail.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/rawmidi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6f30231bdb88..befa9809ff00 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -447,6 +447,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) err = -ENOMEM; goto __error; } + rawmidi_file->user_pversion = 0; init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); while (1) { -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: Fix quirk for Clevo NJ51CU
by Werner Sembach 22 Dec '21

22 Dec '21

The Clevo NJ51CU comes either with the ALC293 or the ALC256 codec, but uses the 0x8686 subproduct id in both cases. The ALC256 codec needs a different quirk for the headset microphone working and and edditional quirk for sound working after suspend and resume. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020 on the ALC256 codec. Setting the value manually makes the sound work again. This patch does this automatically. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Fixes: b5acfe152abaa ("ALSA: hda/realtek: Add some Clove SSID in the ALC293(ALC1220)") Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3599f4c85ebf..3394ef209772 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6546,6 +6546,23 @@ static void alc233_fixup_no_audio_jack(struct hda_codec *codec, alc_process_coef_fw(codec, alc233_fixup_no_audio_jack_coefs); } +static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The Clevo NJ51CU comes either with the ALC293 or the ALC256 codec, + * but uses the 0x8686 subproduct id in both cases. The ALC256 codec + * needs an additional quirk for sound working after suspend and resume. + */ + if (codec->core.vendor_id == 0x10ec0256) { + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x04a11120); + } + else + snd_hda_codec_set_pincfg(codec, 0x1a, 0x04a1113c); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6766,6 +6783,7 @@ enum { ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, + ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME, }; static const struct hda_fixup alc269_fixups[] = { @@ -8490,6 +8508,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_no_audio_jack, }, + [ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_mic_no_presence_and_resume, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8829,7 +8853,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.25.1

2 1

Re: [PATCH 0/4] driver_core: Auxiliary drvdata helper cleanup
by Jason Gunthorpe 22 Dec '21

22 Dec '21

On Tue, Dec 21, 2021 at 04:48:17PM -0800, David E. Box wrote: > On Tue, 2021-12-21 at 20:09 -0400, Jason Gunthorpe wrote: > > On Tue, Dec 21, 2021 at 03:58:48PM -0800, David E. Box wrote: > > > Depends on "driver core: auxiliary bus: Add driver data helpers" patch [1]. > > > Applies the helpers to all auxiliary device drivers using > > > dev_(get/set)_drvdata. Drivers were found using the following search: > > > > > > grep -lr "struct auxiliary_device" $(grep -lr "drvdata" .) > > > > > > Changes were build tested using the following configs: > > > > > > vdpa/mlx5: CONFIG_MLX5_VDPA_NET > > > net/mlx53: CONFIG_MLX5_CORE_EN > > > soundwire/intel: CONFIG_SOUNDWIRE_INTEL > > > RDAM/irdma: CONFIG_INFINIBAND_IRDMA > > > CONFIG_MLX5_INFINIBAND > > > > > > [1] https://www.spinics.net/lists/platform-driver-x86/msg29940.html > > > > I have to say I don't really find this to be a big readability > > improvement. > > I should have referenced the thread [1] discussing the benefit of this change > since the question was asked and answered already. The idea is that drivers > shouldn't have to touch the device API directly if they are already using a > higher level core API (auxiliary bus) that can do that on its behalf. Driver writers should rarely use the auxilary device type directly, the should always immediately container_of it to their proper derived type. > > Also, what use is 'to_auxiliary_dev()' ? I didn't see any users added.. > > This was not added by that patch. It was added by the referenced patch, and seems totally pointless cut and paste, again because nothing should be using the auxiliary_device type for anything more than container_of'ing to their own type. We've been ripping out bus specific APIs in favour of generic ones (see the work on the DMA API for instance) so this whole concept seems regressive, particularly when applied to auxiliary bus which does not have an API of its own. Jason

1 0

[PATCH v2] dt-bindings: dsp: mediatek: add mt8195 dsp document
by YC Hung 22 Dec '21

22 Dec '21

From: "yc.hung" <yc.hung(a)mediatek.com> This patch adds mt8195 dsp document. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v1: Rename yaml file name as mediatek,mt8195-dsp.yaml Refine descriptions for mailbox, memory-region and drop unused labels in examples. --- .../bindings/dsp/mediatek,mt8195-dsp.yaml | 117 ++++++++++++++++++ 1 file changed, 117 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml new file mode 100644 index 000000000000..bde763191d86 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mediatek,mt8195-dsp.yaml @@ -0,0 +1,117 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mediatek,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. + +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + interrupts: + items: + - description: watchdog interrupt + + interrupt-names: + items: + - const: wdt + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: ipc reply between host and audio DSP. + - description: ipc reuqest between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: dma buffer between host and DSP. + - description: DSP system memory. + +required: + - compatible + - reg + - reg-names + - interrupts + - interrupt-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + interrupts = <GIC_SPI 694 IRQ_TYPE_LEVEL_HIGH 0>; + interrupt-names = "wdt"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox0>, <&adsp_mailbox1>; + }; -- 2.18.0

3 2

[PATCH] ASoC: codecs: ak4375: Change invert controls to a stereo switch
by Vincent Knecht 22 Dec '21

22 Dec '21

Don't use enums for DACL/DACR Signal Invert controls, and change them into a stereo "DAC Signal Invert Switch" control. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- sound/soc/codecs/ak4375.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ak4375.c b/sound/soc/codecs/ak4375.c index a893aff42a01..22cda0699341 100644 --- a/sound/soc/codecs/ak4375.c +++ b/sound/soc/codecs/ak4375.c @@ -127,7 +127,6 @@ static DECLARE_TLV_DB_SCALE(hpg_tlv, -4200, 20, 0); static const char * const ak4375_ovolcn_select_texts[] = { "Dependent", "Independent" }; static const char * const ak4375_mdac_select_texts[] = { "x1", "x1/2" }; -static const char * const ak4375_inv_select_texts[] = { "Normal", "Inverting" }; static const char * const ak4375_cpmode_select_texts[] = { "Automatic Switching", "+-VDD Operation", @@ -157,12 +156,6 @@ static const struct soc_enum ak4375_mdacl_enum = static const struct soc_enum ak4375_mdacr_enum = SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 6, ARRAY_SIZE(ak4375_mdac_select_texts), ak4375_mdac_select_texts); -static const struct soc_enum ak4375_invl_enum = - SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 3, - ARRAY_SIZE(ak4375_inv_select_texts), ak4375_inv_select_texts); -static const struct soc_enum ak4375_invr_enum = - SOC_ENUM_SINGLE(AK4375_07_DAC_MONO_MIXING, 7, - ARRAY_SIZE(ak4375_inv_select_texts), ak4375_inv_select_texts); static const struct soc_enum ak4375_cpmode_enum = SOC_ENUM_SINGLE(AK4375_03_POWER_MANAGEMENT4, 2, ARRAY_SIZE(ak4375_cpmode_select_texts), ak4375_cpmode_select_texts); @@ -176,11 +169,11 @@ static const struct snd_kcontrol_new ak4375_snd_controls[] = { SOC_SINGLE_TLV("HP-Amp Analog Volume", AK4375_0D_HP_VOLUME_CONTROL, 0, 0x1f, 0, hpg_tlv), + SOC_DOUBLE("DAC Signal Invert Switch", AK4375_07_DAC_MONO_MIXING, 3, 7, 1, 0), + SOC_ENUM("Digital Volume Control", ak4375_ovolcn_enum), SOC_ENUM("DACL Signal Level", ak4375_mdacl_enum), SOC_ENUM("DACR Signal Level", ak4375_mdacr_enum), - SOC_ENUM("DACL Signal Invert", ak4375_invl_enum), - SOC_ENUM("DACR Signal Invert", ak4375_invr_enum), SOC_ENUM("Charge Pump Mode", ak4375_cpmode_enum), SOC_ENUM("DAC Digital Filter Mode", ak4375_digfil_enum), }; -- 2.31.1

2 1

[PATCH 1/2] ASoC: cs4265: Fix part number ID error message
by Fabio Estevam 22 Dec '21

22 Dec '21

The Chip ID - Register 01h contains the following description as per the CS4265 datasheet: "Bits 7 through 4 are the part number ID, which is 1101b (0Dh)" The current error message is incorrect as it prints CS4265_CHIP_ID, which is the register number, instead of printing the expected part number ID value. To make it clearer, also do a shift by 4, so that the error message would become: [ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index cffd6111afac..b89002189a2b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -611,8 +611,8 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, if (devid != CS4265_CHIP_ID_VAL) { ret = -ENODEV; dev_err(&i2c_client->dev, - "CS4265 Device ID (%X). Expected %X\n", - devid, CS4265_CHIP_ID); + "CS4265 Part Number ID: 0x%x Expected: 0x%x\n", + devid >> 4, CS4265_CHIP_ID_VAL >> 4); return ret; } dev_info(&i2c_client->dev, -- 2.25.1

1 1