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[PATCH v1 0/2] Modify documentation and machine driver for mt8186_rt1019_rt5682s sound card
by Ajye Huang 28 Oct '22

28 Oct '22

v1: - Documentation: Add dmic-gpios optional prop for two DMICs case. - dmic codec driver: - "dmic-gpios" property is used for amixer control to switch the dmic signal source between the Front and Rear Dmic. Thanks for the review! Ajye Huang (2): ASoC: mediatek: dt-bindings: modify machine bindings for two MICs case ASoC: mediatek: mt8186-rt5682: Modify machine driver for two DMICs case .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 6 + .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 104 +++++++++++++++++- 2 files changed, 109 insertions(+), 1 deletion(-) -- 2.25.1

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[PATCH v4 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 28 Oct '22

28 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v4: - replaced readb_poll_timeout() with read_poll_timeout() for the DRSM polling function as the register is u32 wide, not u8 Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 22 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 461 insertions(+), 60 deletions(-) -- 2.25.1

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[PATCH] ASoC: fsl_xcvr: Add Counter registers
by Shengjiu Wang 28 Oct '22

28 Oct '22

These counter registers are part of register list, add them to complete the register map - DMAC counter control registers - Data path Timestamp counter register - Data path bit counter register - Data path bit count timestamp register - Data path bit read timestamp register Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 21 +++++++++++++++++++++ 2 files changed, 61 insertions(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c043efe4548d..2a6802fb2a8b 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -934,6 +934,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_RX_DPTH_CTRL_SET, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_CLR, 0x00002C89 }, { FSL_XCVR_RX_DPTH_CTRL_TOG, 0x00002C89 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_RX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_RX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_SET, 0x00000000 }, { FSL_XCVR_TX_DPTH_CTRL_CLR, 0x00000000 }, @@ -944,6 +952,14 @@ static const struct reg_default fsl_xcvr_reg_defaults[] = { { FSL_XCVR_TX_CS_DATA_3, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_4, 0x00000000 }, { FSL_XCVR_TX_CS_DATA_5, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_SET, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG, 0x00000000 }, + { FSL_XCVR_TX_DPTH_TSCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCTR, 0x00000000 }, + { FSL_XCVR_TX_DPTH_BCRR, 0x00000000 }, { FSL_XCVR_DEBUG_REG_0, 0x00000000 }, { FSL_XCVR_DEBUG_REG_1, 0x00000000 }, }; @@ -975,6 +991,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_RX_DPTH_TSCR: + case FSL_XCVR_RX_DPTH_BCR: + case FSL_XCVR_RX_DPTH_BCTR: + case FSL_XCVR_RX_DPTH_BCRR: case FSL_XCVR_TX_DPTH_CTRL: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: @@ -985,6 +1009,14 @@ static bool fsl_xcvr_readable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: + case FSL_XCVR_TX_DPTH_TSCR: + case FSL_XCVR_TX_DPTH_BCR: + case FSL_XCVR_TX_DPTH_BCTR: + case FSL_XCVR_TX_DPTH_BCRR: case FSL_XCVR_DEBUG_REG_0: case FSL_XCVR_DEBUG_REG_1: return true; @@ -1017,6 +1049,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_RX_DPTH_CTRL_SET: case FSL_XCVR_RX_DPTH_CTRL_CLR: case FSL_XCVR_RX_DPTH_CTRL_TOG: + case FSL_XCVR_RX_DPTH_CNTR_CTRL: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG: case FSL_XCVR_TX_DPTH_CTRL_SET: case FSL_XCVR_TX_DPTH_CTRL_CLR: case FSL_XCVR_TX_DPTH_CTRL_TOG: @@ -1026,6 +1062,10 @@ static bool fsl_xcvr_writeable_reg(struct device *dev, unsigned int reg) case FSL_XCVR_TX_CS_DATA_3: case FSL_XCVR_TX_CS_DATA_4: case FSL_XCVR_TX_CS_DATA_5: + case FSL_XCVR_TX_DPTH_CNTR_CTRL: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_SET: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR: + case FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG: return true; default: return false; diff --git a/sound/soc/fsl/fsl_xcvr.h b/sound/soc/fsl/fsl_xcvr.h index 7f2853c60085..4769b0fca21d 100644 --- a/sound/soc/fsl/fsl_xcvr.h +++ b/sound/soc/fsl/fsl_xcvr.h @@ -49,6 +49,16 @@ #define FSL_XCVR_RX_DPTH_CTRL_CLR 0x188 #define FSL_XCVR_RX_DPTH_CTRL_TOG 0x18c +#define FSL_XCVR_RX_DPTH_CNTR_CTRL 0x1C0 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_SET 0x1C4 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_CLR 0x1C8 +#define FSL_XCVR_RX_DPTH_CNTR_CTRL_TOG 0x1CC + +#define FSL_XCVR_RX_DPTH_TSCR 0x1D0 +#define FSL_XCVR_RX_DPTH_BCR 0x1D4 +#define FSL_XCVR_RX_DPTH_BCTR 0x1D8 +#define FSL_XCVR_RX_DPTH_BCRR 0x1DC + #define FSL_XCVR_TX_DPTH_CTRL 0x220 /* TX datapath ctrl reg */ #define FSL_XCVR_TX_DPTH_CTRL_SET 0x224 #define FSL_XCVR_TX_DPTH_CTRL_CLR 0x228 @@ -59,6 +69,17 @@ #define FSL_XCVR_TX_CS_DATA_3 0x23C #define FSL_XCVR_TX_CS_DATA_4 0x240 #define FSL_XCVR_TX_CS_DATA_5 0x244 + +#define FSL_XCVR_TX_DPTH_CNTR_CTRL 0x260 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_SET 0x264 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_CLR 0x268 +#define FSL_XCVR_TX_DPTH_CNTR_CTRL_TOG 0x26C + +#define FSL_XCVR_TX_DPTH_TSCR 0x270 +#define FSL_XCVR_TX_DPTH_BCR 0x274 +#define FSL_XCVR_TX_DPTH_BCTR 0x278 +#define FSL_XCVR_TX_DPTH_BCRR 0x27C + #define FSL_XCVR_DEBUG_REG_0 0x2E0 #define FSL_XCVR_DEBUG_REG_1 0x2F0 -- 2.34.1

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[PATCH 0/3] Add support for SAI on i.MX93 platform
by Chancel Liu 28 Oct '22

28 Oct '22

This patchset supports SAI on i.MX93 platform. Chancel Liu (3): ASoC: dt-bindings: fsl,sai: Add compatible string for i.MX93 platform ASoC: fsl_sai: Add support for i.MX93 platform ASoC: fsl_sai: Specify the maxburst to 8 on i.MX93 platform .../devicetree/bindings/sound/fsl,sai.yaml | 1 + sound/soc/fsl/fsl_sai.c | 23 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 21 insertions(+), 4 deletions(-) -- 2.25.1

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[PATCH v1] ALSA: hda/realtek: Add quirk for ASUS Zenbook using CS35L41
by Stefan Binding 28 Oct '22

28 Oct '22

This Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using I2C with Internal Boost. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 701a72ec5629a..b4f7ff8cfe41b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9404,6 +9404,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.34.1

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[PATCH 0/6] soundwire: qcom: fixes and support for v1.7
by Srinivas Kandagatla 28 Oct '22

28 Oct '22

This patchset has 2 fixes around fifo handling adds support for v1.7.0 controller found in SM8450 SoCs. Also a patch to make reset optional for v1.6 as this is not required for sc8280xp. Thanks, Srini Srinivas Kandagatla (6): soundwire: qcom: reinit broadcast completion soundwire: qcom: check for outanding writes before doing a read soundwire: qcom: remove unused SWRM_SPECIAL_CMD_ID soundwire: qcom: make reset optional for v1.6 controller dt-bindings: soundwire: qcom: add v1.7.0 support soundwire: qcom: add support for v1.7 Soundwire Controller .../bindings/soundwire/qcom,sdw.txt | 1 + drivers/soundwire/qcom.c | 34 ++++++++++++++++--- 2 files changed, 30 insertions(+), 5 deletions(-) -- 2.21.0

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alsamixer not readable in xterm
by GitHub issues - opened 28 Oct '22

28 Oct '22

alsa-project/alsa-utils issue #176 was opened from Frank-Steiner: Hi, after upgrading from opensuse 15.3 to 15.4 we realized alsamixer is not useable in standard xterm with white background (which is our default) because the black background vanished. Found https://mailman.alsa-project.org/pipermail/alsa-devel/2021-June/185925.html which explains it. But that patch was a bad idea for everyone just using standard xterms and not themed ones. Adding sth. like a "--black-background" switch could be helpful for those people not working with themed X11... cu, Frank Issue URL : https://github.com/alsa-project/alsa-utils/issues/176 Repository URL: https://github.com/alsa-project/alsa-utils

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[RFC][PATCH v2 29/31] timers: ALSA: Use del_timer_shutdown() before freeing timer
by Steven Rostedt 28 Oct '22

28 Oct '22

From: "Steven Rostedt (Google)" <rostedt(a)goodmis.org> Before a timer is freed, del_timer_shutdown() must be called. Link: https://lore.kernel.org/all/20220407161745.7d6754b3@gandalf.local.home/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Austin Kim <austin.kim(a)lge.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Steven Rostedt (Google) <rostedt(a)goodmis.org> --- sound/i2c/other/ak4117.c | 2 +- sound/synth/emux/emux.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 1bc43e927d82..5269ab7321a4 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -47,7 +47,7 @@ static void reg_dump(struct ak4117 *ak4117) static void snd_ak4117_free(struct ak4117 *chip) { - del_timer_sync(&chip->timer); + del_timer_shutdown(&chip->timer); kfree(chip); } diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index a870759d179e..a43025f466bb 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -129,7 +129,7 @@ int snd_emux_free(struct snd_emux *emu) if (! emu) return -EINVAL; - del_timer_sync(&emu->tlist); + del_timer_shutdown(&emu->tlist); snd_emux_proc_free(emu); snd_emux_delete_virmidi(emu); -- 2.35.1

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[PATCH] ASoc: cs42xx8-i2c.c: add module device table for of
by Peter Bergin 28 Oct '22

28 Oct '22

When trying to connect the device with the driver through of it is not working. The of_device_id is defined in cs42xx8.c but is not correctly included in cs42xx8-i2c.c. Also add the matching table for of in the i2c file. Signed-off-by: Peter Bergin <peter(a)berginkonsult.se> --- sound/soc/codecs/cs42xx8-i2c.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c index cb06a06d48b0..6e8ee28d01f8 100644 --- a/sound/soc/codecs/cs42xx8-i2c.c +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -37,6 +37,13 @@ static int cs42xx8_i2c_remove(struct i2c_client *i2c) return 0; } +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); + static struct i2c_device_id cs42xx8_i2c_id[] = { {"cs42448", (kernel_ulong_t)&cs42448_data}, {"cs42888", (kernel_ulong_t)&cs42888_data}, -- 2.34.1

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[PATCH v2] ASoC: dt-bindings: Document audio OF graph dai-tdm-slot-num dai-tdm-slot-width props
by Marek Vasut 27 Oct '22

27 Oct '22

Document dai-tdm-slot-num and dai-tdm-slot-width props as those are parsed by simple graph card and may therefore appear in audio OF graph node. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: devicetree(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org To: linux-arm-kernel(a)lists.infradead.org --- V2: Drop the definition refs and just fill in type and description --- .../devicetree/bindings/sound/audio-graph-port.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index bc46a95ed8400..64654ceef2089 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -74,6 +74,12 @@ patternProperties: convert-sample-format: $ref: "/schemas/sound/dai-params.yaml#/$defs/dai-sample-format" + dai-tdm-slot-num: + description: Number of slots in use. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: Width in bits for each slot. + $ref: /schemas/types.yaml#/definitions/uint32 dai-tdm-slot-width-map: description: Mapping of sample widths to slot widths. For hardware that cannot support a fixed slot width or a slot width always -- 2.35.1

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