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[PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58381801e41d4..4571785736f76 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

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Re: [PATCH] ALSA: hda: cs35l56: Don't use the device index as a calibration index
by Takashi Iwai 21 Aug '24

21 Aug '24

On Wed, 21 Aug 2024 14:47:11 +0200, Simon Trimmer wrote: > > The HDA driver cannot assume that the order that the devices are > specified in the cirrus,dev-index matches the order of calibration > entries. > > Only a calibration entry with a matching silicon id will be used. > > Fixes: cfa43aaa7948 ("ALSA: hda: cs35l56: Apply amp calibration from EFI data") > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Thanks, applied now. Takashi

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Re: [RESEND] ASoC: mediatek: mt8188-mt6359: Modify key
by Mark Brown 21 Aug '24

21 Aug '24

On Wed, Aug 21, 2024 at 03:40:51PM +0800, Zhang Yi wrote: > In order to get the correct keys when using the ES8326.We will associate > SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN > when the ES8326 flag is recognized. Please please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review.

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[PATCH v2 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 20 Aug '24

20 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org> Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- Change from v1 to v2 - fix MAINTAINERS --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- MAINTAINERS | 2 +- 3 files changed, 56 insertions(+), 28 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; diff --git a/MAINTAINERS b/MAINTAINERS index 118d9cf64e9cb..4ced49519930b 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22713,12 +22713,12 @@ F: Documentation/devicetree/bindings/sound/tas2552.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.34.1

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[PATCH] ASoC: amd: acp: replace desc->rev check with acp pci revision id
by Vijendar Mukunda 20 Aug '24

20 Aug '24

Replace acp descriptor structure member 'rev' check with acp pci revision id. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 0f6115c8b005..e4d46fdda88b 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -236,7 +236,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, unsigned int image_length) { struct snd_sof_dev *sdev = adata->dev; - const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int tx_count, fw_qualifier, val; int ret; @@ -265,8 +264,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* psp_send_cmd only required for vangogh platform */ + if (adata->pci_rev == ACP_VANGOGH_PCI_ID && + !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { /* Modify IRAM and DRAM size */ ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); if (ret) @@ -285,8 +285,8 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for renoir platform (rev - 3) */ - if (desc->rev == 3) { + /* psp_send_cmd only required for renoir platform*/ + if (adata->pci_rev == ACP_RN_PCI_ID) { ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); if (ret) return ret; @@ -405,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_sw0_i2s_err_reason, 0); /* ACP_SW1_I2S_ERROR_REASON is newly added register from rmb platform onwards */ - if (desc->rev >= 6) + if (adata->pci_rev >= ACP_RMB_PCI_ID) snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SW1_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; @@ -431,6 +431,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) static int acp_power_on(struct snd_sof_dev *sdev) { const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *adata = sdev->pdata->hw_pdata; unsigned int base = desc->pgfsm_base; unsigned int val; unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; @@ -441,13 +442,14 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - switch (desc->rev) { - case 3: - case 5: + switch (adata->pci_rev) { + case ACP_RN_PCI_ID: + case ACP_VANGOGH_PCI_ID: acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; break; - case 6: + case ACP_RMB_PCI_ID: + case ACP63_PCI_ID: acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; break; -- 2.34.1

2 1

[RFC PATCH 00/16] soundwire/ASoC: speed-up downloads with BTP/BRA protocol
by Pierre-Louis Bossart 20 Aug '24

20 Aug '24

This RFC patchset suggests a new API for ASoC codec drivers to use for firmware/table downloads. Problem statement: All existing transfers initiated by codec drivers rely on SoundWire read/write commands, which can only support ONE byte per frame. With the typical 48kHz frame rate, this means 384 kbits/s. In addition, the command/control is typically handled with a FIFO and interrupts which adds more software overhead. To give a practical reference, sending 32Kb takes 2.5s on Intel platforms, which means about 105kbit/s only. Additional non-audio activity is likely to adversely impact interrupt scheduling and further decrease the transfer speeds. New SDCA-based codecs have a need to download tables and DSP firmware which are typically between 20 and 256 Kb. The slow bus operation has a direct impact on boot/resume times, and clearly having to wait more than 300ms is a showstopper in terms of latency requirements and user-experience. Suggested solution: The MIPI specification and most of the new codecs support the Bulk Transfer Protocol (BTP) and specifically the Bulk Register Access (BRA) configuration. This mode reclaims the 'audio' data space of the SoundWire frame to send firmware/coefficients over the DataPort 0 (DP0). The API suggested is rather simple, with the following sequence expected: open(): reserve resources and prepare hardware send_async(): trigger DMAs and perform SoundWire bank switch wait(): wait for DMA completion and disable DMAs close(): release resources Benefits: Even after accounting for the protocol overhead, the data can be sent 8x or 16x faster on the link than with the regular commands. With the use of DMAs, the software overhead becomes limited to the initialization. Measured results show that transferring the same 32Kb takes about 100ms, a 25x improvement on the baseline write() commands with an actual bitrate of 2.6 Mbits/s. These results are a measure of bus/hardware performance mainly, and will typically not be too modified by the CPU activity and scheduling. The performance for reads is similar, with a 25x speedup measured. Limitations: Setting up the transfers over DP0 takes time, and the reliance on DMAs on the host side brings alignment restrictions. The BTP/BRA protocol is really only relevant for "large" transfers done during boot/resume *before* audio transfers take place. Mixing BTP/BRA and audio is a nightmare, this patchset suggests a mutual-exclusion between two usages. Scope: This patchset only exposes the API and a debugfs interface to initiate commands, validate results and measure performance. The actual use of the API is left as an exercise for codec driver developers. This patchset depends on a number of pre-requisite patches and will not build on top of any for-next branches. The main intent of this RFC is to gather comments on the usage, API, benefits and restrictions. The code and functionality was tested on an Intel LunarLake RVP platform connected to a Realtek RT711-SDCA device. Acknowledgements: Thanks to Zeek Tsai at Realtek for providing test sequences that helped reconcile the data formatted by the host driver with the expected results on the code side. Pierre-Louis Bossart (16): Documentation: driver: add SoundWire BRA description soundwire: cadence: add BTP support for DP0 soundwire: stream: extend sdw_alloc_stream() to take 'type' parameter soundwire: extend sdw_stream_type to BPT soundwire: stream: special-case the bus compute_params() routine soundwire: stream: reuse existing code for BPT stream soundwire: bus: add API for BPT protocol soundwire: bus: add bpt_stream pointer soundwire: crc8: add constant table soundwire: cadence: add BTP/BRA helpers to format data soundwire: intel_auxdevice: add indirection for BPT open/close/send_async/wait ASoC: SOF: Intel: hda-sdw-bpt: add helpers for SoundWire BPT DMA soundwire: intel: add BPT context definition soundwire: intel_ace2x: add BPT open/close/send_async/wait soundwire: debugfs: add interface for BPT/BRA transfers ASoC: rt711-sdca: add DP0 support Documentation/driver-api/soundwire/bra.rst | 478 +++++++++++++ Documentation/driver-api/soundwire/index.rst | 1 + Documentation/driver-api/soundwire/stream.rst | 2 +- .../driver-api/soundwire/summary.rst | 5 +- drivers/soundwire/Kconfig | 1 + drivers/soundwire/Makefile | 4 +- drivers/soundwire/amd_manager.c | 2 +- drivers/soundwire/bus.c | 77 +++ drivers/soundwire/bus.h | 18 + drivers/soundwire/cadence_master.c | 646 +++++++++++++++++- drivers/soundwire/cadence_master.h | 30 + drivers/soundwire/crc8.c | 277 ++++++++ drivers/soundwire/crc8.h | 11 + drivers/soundwire/debugfs.c | 122 +++- .../soundwire/generic_bandwidth_allocation.c | 84 ++- drivers/soundwire/intel.h | 12 + drivers/soundwire/intel_ace2x.c | 377 ++++++++++ drivers/soundwire/intel_auxdevice.c | 55 ++ drivers/soundwire/qcom.c | 2 +- drivers/soundwire/stream.c | 137 +++- include/linux/soundwire/sdw.h | 91 ++- include/linux/soundwire/sdw_intel.h | 16 + include/sound/hda-sdw-bpt.h | 76 +++ sound/soc/codecs/rt711-sdca-sdw.c | 8 + sound/soc/qcom/sdw.c | 2 +- sound/soc/sof/intel/Kconfig | 8 +- sound/soc/sof/intel/Makefile | 4 + sound/soc/sof/intel/hda-sdw-bpt.c | 328 +++++++++ 28 files changed, 2810 insertions(+), 64 deletions(-) create mode 100644 Documentation/driver-api/soundwire/bra.rst create mode 100644 drivers/soundwire/crc8.c create mode 100644 drivers/soundwire/crc8.h create mode 100644 include/sound/hda-sdw-bpt.h create mode 100644 sound/soc/sof/intel/hda-sdw-bpt.c -- 2.39.2

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[PATCH] ASoC: dt-bindings: samsung,odroid: drop stale clocks
by Krzysztof Kozlowski 20 Aug '24

20 Aug '24

Clocks property was present only to allow usage of assigned-clocks in the sound card node, however in upstream DTS the assigned-clocks were moved in commit 4afb06afd768 ("ARM: dts: exynos: move assigned-clock* properties to i2s0 node in Odroid XU4") to respective I2S nodes. Linux drivers never parsed "clocks" so it can be safely dropped. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/samsung,odroid.yaml | 5 ----- 1 file changed, 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index b77284e3e26a..c3dea852cc8d 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -27,11 +27,6 @@ properties: - const: samsung,odroid-xu4-audio deprecated: true - assigned-clock-parents: true - assigned-clock-rates: true - assigned-clocks: true - clocks: true - cpu: type: object additionalProperties: false -- 2.43.0

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[PATCH alsa-lib] pcm: implement snd_pcm_hw_params_get_sync() and obsolete snd_pcm_info_get_sync()
by Jaroslav Kysela 20 Aug '24

20 Aug '24

Use the new clock source mechanism to get information about similar PCM clock sources for PCM streams. Link: https://lore.kernel.org/linux-sound/20240625172836.589380-1-perex@perex.cz/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/pcm.h | 4 ++++ include/sound/uapi/asound.h | 3 ++- src/pcm/pcm.c | 30 ++++++++++++++++++++++++++---- 3 files changed, 32 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 102ff812..b5c67889 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -508,6 +508,9 @@ typedef union _snd_pcm_sync_id { unsigned int id32[4]; } snd_pcm_sync_id_t; +/** synchronization ID size (see snd_pcm_hw_params_get_sync) */ +#define SND_PCM_HW_PARAMS_SYNC_SIZE 16 + /** Infinite wait for snd_pcm_wait() */ #define SND_PCM_WAIT_INFINITE (-1) /** Wait for next i/o in snd_pcm_wait() */ @@ -747,6 +750,7 @@ int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_den); int snd_pcm_hw_params_get_sbits(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params); +const unsigned char * snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params); #if 0 typedef struct _snd_pcm_hw_strategy snd_pcm_hw_strategy_t; diff --git a/include/sound/uapi/asound.h b/include/sound/uapi/asound.h index f3b2b94d..012fd890 100644 --- a/include/sound/uapi/asound.h +++ b/include/sound/uapi/asound.h @@ -426,7 +426,8 @@ struct snd_pcm_hw_params { unsigned int rate_num; /* R: rate numerator */ unsigned int rate_den; /* R: rate denominator */ snd_pcm_uframes_t fifo_size; /* R: chip FIFO size in frames */ - unsigned char reserved[64]; /* reserved for future */ + unsigned char sync[16]; /* R: synchronization ID (perfect sync - one clock source) */ + unsigned char reserved[48]; /* reserved for future */ }; enum { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 62e76e7e..f8581191 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -494,8 +494,8 @@ software parameter. There are two functions allowing link multiple streams together. In the case, the linking means that all operations are synchronized. Because the drivers cannot guarantee the synchronization (sample resolution) on hardware -lacking this feature, the #snd_pcm_info_get_sync() function -returns synchronization ID - #snd_pcm_sync_id_t, which is equal +lacking this feature, the #snd_pcm_hw_params_get_sync() function +returns 16-byte synchronization ID, which is equal for hardware synchronized streams. When the #snd_pcm_link() function is called, all operations managing the stream state for these two streams are joined. The opposite function is #snd_pcm_unlink(). @@ -3948,6 +3948,25 @@ int snd_pcm_hw_params_get_fifo_size(const snd_pcm_hw_params_t *params) return params->fifo_size; } +/** + * \brief Get hardware synchronization ID from a PCM info container + * \param params Configuration space + * \return 16-byte synchronization ID (use #SND_PCM_HW_PARAMS_SYNC_SIZE) + * + * This synchronization ID determines the similar clocks for the + * PCM stream between multiple devices (including different cards). + * "All zeros" means "not set". The contents of the ID can be used + * only for a comparison with the contents of another ID returned + * from this function. Applications should not do a comparison with + * hard-coded values, because the implementation generating such + * synchronization IDs may be changed in future. + */ +const unsigned char *snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t *params) +{ + assert(params); + return params->sync; +} + /** * \brief Fill params with a full configuration space for a PCM * \param pcm PCM handle @@ -7332,7 +7351,7 @@ unsigned int snd_pcm_info_get_subdevices_avail(const snd_pcm_info_t *obj) } /** - * \brief Get hardware synchronization ID from a PCM info container + * \brief (DEPRECATED) Get hardware synchronization ID from a PCM info container * \param obj PCM info container * \return hardware synchronization ID */ @@ -7340,9 +7359,12 @@ snd_pcm_sync_id_t snd_pcm_info_get_sync(const snd_pcm_info_t *obj) { snd_pcm_sync_id_t res; assert(obj); - memcpy(&res, &obj->sync, sizeof(res)); + bzero(&res, sizeof(res)); return res; } +#ifndef DOC_HIDDEN +link_warning(snd_pcm_info_get_sync, "Warning: snd_pcm_info_get_sync is deprecated, consider to use snd_pcm_hw_params_get_sync"); +#endif /** * \brief Set wanted device inside a PCM info container (see #snd_ctl_pcm_info) -- 2.45.2

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[PATCH v2 00/14] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. Changes in v2: - added two new patches (leading the series) - the first moves all device entries previously under the skylake-driver dependency under the avs-driver - the second patch drops struct skl_machine_pdata usage to fix compilation errors as pointed out by IKP - no further changes, retained Acked-by tags for these [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (14): ALSA: hda: Move SST device entries to AVS ASoC: Intel: Drop skl_machine_pdata usage ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/hda/intel-dsp-config.c | 111 +- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- .../intel/common/soc-acpi-intel-cnl-match.c | 6 - .../intel/common/soc-acpi-intel-ehl-match.c | 1 - .../intel/common/soc-acpi-intel-hda-match.c | 6 - .../intel/common/soc-acpi-intel-icl-match.c | 6 - .../intel/common/soc-acpi-intel-kbl-match.c | 11 - .../intel/common/soc-acpi-intel-skl-match.c | 5 - sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 46 files changed, 59 insertions(+), 22871 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

4 19

[PATCH 00/12] ASoC: Intel: Remove skylake driver
by Cezary Rojewski 20 Aug '24

20 Aug '24

A spiritual successor to haswell/baytrail removal series [1]. The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. The skylake-driver related UAPI has been removed with "ASoC: Drop soc-topology ABI v4 support" [2]. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume this DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. [1]: https://lore.kernel.org/alsa-devel/20201006064907.16277-1-cezary.rojewski@i… [2]: https://lore.kernel.org/alsa-devel/20240403091629.647267-1-cezary.rojewski@… Cezary Rojewski (12): ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 152 +- sound/soc/intel/boards/Makefile | 20 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------ sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ----- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ---- .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 --- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 ------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 -- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ----- sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 -- sound/soc/intel/skylake/skl-sst.c | 599 --- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------ sound/soc/intel/skylake/skl.h | 207 - 39 files changed, 9 insertions(+), 22775 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h -- 2.25.1

3 14

[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 20 Aug '24

20 Aug '24

Between the kernel releases 6.9.9 and 6.10.0, something changed that causes heavy digital distortion on playback from at least ALSA and PipeWire. The rhythm of the music is discernable in the output, so it isn't just random noise. It sounds a bit like bitcrushing, but not quite. The generated overtones appear to be dependet on the sample rate. I am sorry I can not be more specific, there are no kernel messages this time. As Mr. Sakamoto recently committed all these changes to firewire he might be able to identify the issue easily, which is why I'd like to ask him to look into it. Kind regards, Edmund Raile. #regzbot introduced: v6.9.9..v6.10

2 1

[PATCH v1] ASoc: tas2781: fixed the issue that the chip do not shutdown immediatly after aplay stopped
by Shenghao Ding 20 Aug '24

20 Aug '24

Issue reported from customer that the chip do not shutdown after aplay stopped until 6 mins later. Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Drop tasdevice_dapm_event and implement .stream_mute in the tasdevice_dai_ops - Remove unnecessary line feed for tasdevice_codec_remove --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..145c45349399 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -557,30 +557,22 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; - /* Codec Lock Hold */ - mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); - /* Codec Lock Release*/ - mutex_unlock(&tas_priv->codec_lock); + /* Codec Lock/UnLock */ + guard(mutex)(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, mute); return 0; } static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), SND_SOC_DAPM_INPUT("DMIC") }; @@ -667,6 +659,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { -- 2.34.1

3 3

[PATCH v7 2/2] ARM: dts: imx6: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi | 15 ++++++++++++--- arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi | 9 +++++++-- .../boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts | 9 +++++++-- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 9 +++++++-- 12 files changed, 104 insertions(+), 28 deletions(-) diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts index 95b49fc83f7b..299106fbe51c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-cm-fx6.dts @@ -127,12 +127,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts index a7d5693c5ab7..8d2b608e0b90 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-prti6q.dts @@ -111,12 +111,21 @@ simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts index 7c298d9aa21e..5353a0c24420 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6q-tbs2910.dts @@ -90,11 +90,16 @@ sound-sgtl5000 { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi index ea40623d12e5..edf55760a5c1 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apalis.dtsi @@ -197,11 +197,20 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi index 3a46ade3b6bd..9e97ef5e43f2 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-apf6dev.dtsi @@ -121,11 +121,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi index d3a7a6eeb8e0..b01670cdd52c 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-colibri.dtsi @@ -142,12 +142,21 @@ sound { ssi-controller = <&ssi1>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + /* Optional S/PDIF in on SODIMM 88 and out on SODIMM 90, 137 or 168 */ sound_spdif: sound-spdif { compatible = "fsl,imx-audio-spdif"; - spdif-controller = <&spdif>; - spdif-in; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>, <&spdif_in>; model = "imx-spdif"; status = "disabled"; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi index 761566ae3cf5..bd66430c1d78 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-cubox-i.dtsi @@ -100,12 +100,17 @@ v_usb1: regulator-v-usb1 { vin-supply = <&v_5v0>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "Integrated SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; gpio-keys { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi index a955c77cd499..d1ad65ab6b72 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-hummingboard.dtsi @@ -140,12 +140,17 @@ sound_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "On-board SPDIF"; /* IMX6 doesn't implement this yet */ - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi index 0a3deaf92eea..55fb7b904220 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-sabreauto.dtsi @@ -143,12 +143,17 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-sabreauto-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; backlight { diff --git a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi index 38abb6b50f6c..7130b9c3b3aa 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6qdl-wandboard.dtsi @@ -26,11 +26,16 @@ sound { mux-ext-port = <3>; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; reg_1p5v: regulator-1p5v { diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts index b0c27b9b0244..dfbfb8119bf3 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sabreauto.dts @@ -97,11 +97,16 @@ sound-cs42888 { "AIN2R", "Line In Jack"; }; + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-in; + audio-cpu = <&spdif>; + audio-codec = <&spdif_in>; }; }; diff --git a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi index 7d4170c27732..277a6e039045 100644 --- a/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi +++ b/arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi @@ -183,12 +183,17 @@ panel_in: endpoint { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx6sx-sdb-spdif", "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; + audio-cpu = <&spdif>; + audio-codec = <&spdif_out>; }; }; -- 2.34.1

1 0

[PATCH v7 1/2] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 20 Aug '24

20 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 930e14fec423..5f8336217bb8 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..33d73f3dc187 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..a87d0692c3bb 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + compatible = "linux,spdif-dit"; + #sound-dai-cells = <0>; + }; + + spdif_in: spdif-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + compatible = "linux,spdif-dir"; + #sound-dai-cells = <0>; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

1 0

[PATCH] ASoc: tas2781: replace devm_kzalloc and scnprintf with devm_kasprintf
by Shenghao Ding 20 Aug '24

20 Aug '24

Replace devm_kzalloc and scnprintf with devm_kasprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..1451ba0fa5b4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Profile Id"); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); - if (!prog_name || !conf_name) { + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, + "Speaker Program Id"); + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kasprintf(tas_priv->dev, GFP_KERNEL, "Speaker Config Id"); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] soundwire: qcom: Use devm_clk_get_enabled() helpers
by Huan Yang 20 Aug '24

20 Aug '24

The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: Huan Yang <link(a)vivo.com> --- drivers/soundwire/qcom.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2a57ed9464b4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1527,24 +1527,22 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (IS_ERR(ctrl->audio_cgcr)) { dev_err(dev, "Failed to get cgcr reset ctrl required for SW gating\n"); ret = PTR_ERR(ctrl->audio_cgcr); - goto err_init; + return ret; } } ctrl->irq = of_irq_get(dev->of_node, 0); if (ctrl->irq < 0) { ret = ctrl->irq; - goto err_init; + return ret; } - ctrl->hclk = devm_clk_get(dev, "iface"); + ctrl->hclk = devm_clk_get_enabled(dev, "iface"); if (IS_ERR(ctrl->hclk)) { ret = dev_err_probe(dev, PTR_ERR(ctrl->hclk), "unable to get iface clock\n"); - goto err_init; + return ret; } - clk_prepare_enable(ctrl->hclk); - ctrl->dev = dev; dev_set_drvdata(&pdev->dev, ctrl); mutex_init(&ctrl->port_lock); @@ -1558,7 +1556,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) ret = qcom_swrm_get_port_config(ctrl); if (ret) - goto err_clk; + return ret; params = &ctrl->bus.params; params->max_dr_freq = DEFAULT_CLK_FREQ; @@ -1586,7 +1584,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "soundwire", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire irq\n"); - goto err_clk; + return ret; } ctrl->wake_irq = of_irq_get(dev->of_node, 1); @@ -1597,7 +1595,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) "swr_wake_irq", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire wake irq\n"); - goto err_init; + return ret; } } @@ -1612,7 +1610,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", ret); - goto err_clk; + return ret; } qcom_swrm_init(ctrl); @@ -1642,9 +1640,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) err_master_add: sdw_bus_master_delete(&ctrl->bus); -err_clk: - clk_disable_unprepare(ctrl->hclk); -err_init: return ret; } @@ -1653,7 +1648,6 @@ static void qcom_swrm_remove(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(&pdev->dev); sdw_bus_master_delete(&ctrl->bus); - clk_disable_unprepare(ctrl->hclk); } static int __maybe_unused swrm_runtime_resume(struct device *dev) -- 2.45.2

1 0

[PATCH v3 0/8] MSM8953/MSM8976 ASoC support
by Adam Skladowski 20 Aug '24

20 Aug '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Changes since v2 =============== 1. Guarded ioremap on quinary mux with checking if resource is not NULL 2. Introduced compatible based reg overriding in wcd-analog driver 3. Separated PM8950/PM8953 codec patches 4. Moved reg/reg-names back into top-level and rewrote as in comments 5. Separated MSM8953/MSM8976 QDSP6 patches 6. Added Lineout patch [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Signed-off-by: Adam Skladowski <a39.skl(a)gmail.com> --- Adam Skladowski (4): ASoC: dt-bindings: pm8916-wcd-analog-codec: Document pm8950/pm8953 ASoC: msm8916-wcd-analog: add pm8950 codec ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard ASoC: qcom: apq8016_sbc: Add support for msm8976 SoC Vladimir Lypak (4): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add pm8953 codec ASoC: qcom: apq8016_sbc: Add support for msm8953 SoC ASoC: msm8916-wcd-analog: add lineout output .../bindings/sound/qcom,apq8016-sbc-sndcard.yaml | 27 ++++ .../sound/qcom,pm8916-wcd-analog-codec.yaml | 6 +- sound/soc/codecs/msm8916-wcd-analog.c | 145 ++++++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 88 ++++++++++++- 4 files changed, 257 insertions(+), 9 deletions(-) --- base-commit: 31c2a5c4bf3673b0bf098f358ea931b9ada1e7de change-id: 20240729-msm8953-msm8976-asoc-65f973a13165 Best regards, -- Adam Skladowski <a39.skl(a)gmail.com>

6 21

[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

discussion: why is the mono profile in ALSA not autogenerated by default?
by xeroxslayer＠proton.me 20 Aug '24

20 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

1 0

[RFC PATCH v2 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 20 Aug '24

20 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: reserve space in snd_compr_metadata.key for private usage ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 791 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1297 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 13

[PATCH alsa-utils] aseqdump: Add missing dump of UMP Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The handling of Set Key Signature message of the Flex Data type was missing by some reason. Now the definition was added in alsa-lib ump_msg.h, so let's show the contents. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- seq/aseqdump/aseqdump.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/seq/aseqdump/aseqdump.c b/seq/aseqdump/aseqdump.c index 1d9c01e01dd3..e37c87b45fc1 100644 --- a/seq/aseqdump/aseqdump.c +++ b/seq/aseqdump/aseqdump.c @@ -964,6 +964,14 @@ static void dump_ump_flex_data_event(const unsigned int *ump) return; } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_KEY_SIGNATURE) { + printf("UMP Set Key Signature sharps/flats %d, tonic %d\n", + fh->set_key_sig.sharps_flats, + fh->set_key_sig.tonic_note); + return; + } + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_CHORD_NAME) { printf("UMP Set Chord Name tonic %d %d %d, alt1 %d/%d, alt2 %d/%d, alt3 %d/%d, alt4 %d/%d, bass %d %d %d, alt1 %d/%d alt2 %d/%d\n", -- 2.43.0

1 0

[PATCH alsa-lib] ump_msg: Add missing definition for Set Key Signature Message
by Takashi Iwai 20 Aug '24

20 Aug '24

The Set Key Signature message was missing at the time we defined for Flex Data types. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/ump_msg.h | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/include/ump_msg.h b/include/ump_msg.h index a64cb317889e..5f0a45acd44c 100644 --- a/include/ump_msg.h +++ b/include/ump_msg.h @@ -606,6 +606,39 @@ typedef struct _snd_ump_msg_set_metronome { #endif } __attribute((packed)) snd_ump_msg_set_metronome_t; +/** Set Key Signature Message (128bit) */ +typedef struct _snd_ump_msg_set_key_sig { +#ifdef SNDRV_BIG_ENDIAN_BITFIELD + uint8_t type:4; /**< UMP packet type */ + uint8_t group:4; /**< UMP Group */ + uint8_t format:2; /**< Format */ + uint8_t addrs:2; /**< Address */ + uint8_t channel:4; /**< Channel */ + uint8_t status_bank; /**< Status Bank */ + uint8_t status; /**< Status */ + + uint8_t sharps_flats:4; /**< Sharps/Flats */ + uint8_t tonic_note:4; /**< Tonic Note 1 */ + uint8_t reserved1[3]; /**< Unused */ + + uint32_t reserved2[2]; /**< Unused */ +#else + uint8_t status; /**< Status */ + uint8_t status_bank; /**< Status Bank */ + uint8_t channel:4; /**< Channel */ + uint8_t addrs:2; /**< Address */ + uint8_t format:2; /**< Format */ + uint8_t group:4; /**< UMP Group */ + uint8_t type:4; /**< UMP packet type */ + + uint8_t reserved1[3]; /**< Unused */ + uint8_t tonic_note:4; /**< Tonic Note */ + uint8_t sharps_flats:4; /**< Sharps/Flats */ + + uint32_t reserved2[2]; /**< Unused */ +#endif +} __attribute((packed)) snd_ump_msg_set_key_sig_t; + /** Set Chord Name Message (128bit) */ typedef struct _snd_ump_msg_set_chord_name { #ifdef SNDRV_BIG_ENDIAN_BITFIELD @@ -678,6 +711,7 @@ typedef union _snd_ump_msg_flex_data { snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ snd_ump_msg_set_time_sig_t set_time_sig; /**< Set Time Signature */ snd_ump_msg_set_metronome_t set_metronome; /**< Set Metronome */ + snd_ump_msg_set_key_sig_t set_key_sig; /**< Set Key Signature */ snd_ump_msg_set_chord_name_t set_chord_name; /**< Set Chord Name */ snd_ump_msg_hdr_t hdr; /**< UMP header */ uint32_t raw[4]; /**< raw UMP packet */ -- 2.43.0

1 0

[PATCH 1/1] ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml
by Frank Li 19 Aug '24

19 Aug '24

Convert binding doc tlv320aic31xx.txt to yaml format. Additional change: - add i2c node in example. - replace MICBIAS_OFF with MICBIAS_2_0v in example because MICBIAS_OFF have been defined in header file. - add ref to dai-common.yaml. - add #sound-dai-cells. Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a30000/codec@18: failed to match any schema with compatible: ['ti,tlv320dac3100'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tlv320dac3100.yaml | 127 ++++++++++++++++++ .../bindings/sound/tlv320aic31xx.txt | 77 ----------- 2 files changed, 127 insertions(+), 77 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 0000000000000..85e937e34962d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding <shenghao-ding(a)ti.com> + +description: | + CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + + CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + + The pins can be used in referring sound node's audio-routing property. + +properties: + compatible: + enum: + - ti,tlv320aic310x # - Generic TLV320AIC31xx with mono speaker amp + - ti,tlv320aic311x # - Generic TLV320AIC31xx with stereo speaker amp + - ti,tlv320aic3100 # - TLV320AIC3100 (mono speaker amp, no MiniDSP) + - ti,tlv320aic3110 # - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + - ti,tlv320aic3120 # - TLV320AIC3120 (mono speaker amp, MiniDSP) + - ti,tlv320aic3111 # - TLV320AIC3111 (stereo speaker amp, MiniDSP) + - ti,tlv320dac3100 # - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + - ti,tlv320dac3101 # - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320aic31xx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_2_0V>; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <&regulator>; + SPRVDD-supply = <&regulator>; + SPLVDD-supply = <&regulator>; + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt deleted file mode 100644 index bbad98d5b9862..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ /dev/null @@ -1,77 +0,0 @@ -Texas Instruments - tlv320aic31xx Codec module - -The tlv320aic31xx serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp - "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp - "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) - "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) - "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) - "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) - "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) - -- reg - <int> - I2C slave address -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai31xx-micbias-vg - MicBias Voltage setting - 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V - 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V - 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD - If this node is not mentioned or if the value is unknown, then - micbias is set to 2.0V. -- ai31xx-ocmv - output common-mode voltage setting - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * HPL - * HPR - * SPL, devices with stereo speaker amp - * SPR, devices with stereo speaker amp - * SPK, devices with mono speaker amp - * MICBIAS - -CODEC input pins: - * MIC1LP, devices with ADC - * MIC1RP, devices with ADC - * MIC1LM, devices with ADC - * AIN1, devices without ADC - * AIN2, devices without ADC - -The pins can be used in referring sound node's audio-routing property. - -Example: -#include <dt-bindings/gpio/gpio.h> -#include <dt-bindings/sound/tlv320aic31xx.h> - -tlv320aic31xx: tlv320aic31xx@18 { - compatible = "ti,tlv320aic311x"; - reg = <0x18>; - - ai31xx-micbias-vg = <MICBIAS_OFF>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - HPVDD-supply = <&regulator>; - SPRVDD-supply = <&regulator>; - SPLVDD-supply = <&regulator>; - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; -- 2.34.1

3 2

Re: [PATCH] ALSA: hda/realtek: Convert existing CS35L56 products to use autodetect fixup function
by Takashi Iwai 19 Aug '24

19 Aug '24

On Mon, 19 Aug 2024 14:37:36 +0200, Simon Trimmer wrote: > > The existing CS35L56 products can make use of the fixup function that > works out the component binding details so we can remove the fixed > configuration fixup functions. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Looks like a nice cleanup. Applied to for-next branch now. Thanks. Takashi

1 0

[PATCH v6 6/7] arm64: dts: imx8m: update spdif sound card node properties
by Elinor Montmasson 19 Aug '24

19 Aug '24

The merge of imx-spdif driver into fsl-asoc-card brought new DT properties that can be used with the "fsl,imx-audio-spdif" compatible: * The "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an S/PDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". While backward compatibility is kept to support properties "spdif-controller", "spdif-in" and "spdif-out", using new properties has several benefits: * "audio-cpu" and "audio-codec" are more generic names reflecting that the fsl-asoc-card driver supports multiple hardware. They are properties already used by devices using the fsl-asoc-card driver. They are also similar to properties of simple-card: "cpu" and "codec". * "spdif-in" and "spdif-out" imply the use of the dummy codec in the driver. However, there are already two codec drivers for the S/PDIF, spdif_transmitter and spdif_receiver. It is better to declare S/PDIF Tx and Rx devices in a DT, and then reference them with "audio-codec" than using the dummy codec. For those reasons, this commit updates in-tree DTs to use the new properties: * Rename "spdif-controller" property to "audio-cpu". * Declare S/PDIF transmitter and/or receiver devices, and use them with the "audio-codec" property instead of "spdif-out" and/or "spdif-in". These modifications were tested only on an imx8mn-evk board. Note that out-of-tree and old DTs are still supported. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi | 15 +++++++++--- arch/arm64/boot/dts/freescale/imx8mq-evk.dts | 24 +++++++++++++++---- 3 files changed, 43 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi index 90d1901df2b1..348855a41852 100644 --- a/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mm-evk.dtsi @@ -180,12 +180,21 @@ cpu { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; }; diff --git a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi index 9e0259ddf4bc..6a47e09703a7 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi +++ b/arch/arm64/boot/dts/freescale/imx8mn-evk.dtsi @@ -124,12 +124,21 @@ sound-wm8524 { "Line Out Jack", "LINEVOUTR"; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; }; sound-micfil { diff --git a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts index 7507548cdb16..b953865f0b46 100644 --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -125,19 +125,33 @@ link_codec: simple-audio-card,codec { }; }; + spdif_out: spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + + spdif_in: spdif-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; + }; + sound-spdif { compatible = "fsl,imx-audio-spdif"; model = "imx-spdif"; - spdif-controller = <&spdif1>; - spdif-out; - spdif-in; + audio-cpu = <&spdif1>; + audio-codec = <&spdif_out>, <&spdif_in>; + }; + + hdmi_arc_in: hdmi-arc-in { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dir"; }; sound-hdmi-arc { compatible = "fsl,imx-audio-spdif"; model = "imx-hdmi-arc"; - spdif-controller = <&spdif2>; - spdif-in; + audio-cpu = <&spdif2>; + audio-codec = <&hdmi_arc_in>; }; }; -- 2.34.1

2 2

[PATCH v11 00/10] spi: Add support for stacked/parallel memories
by Amit Kumar Mahapatra 19 Aug '24

19 Aug '24

This patch series updated the spi-nor, spi core and the AMD-Xilinx GQSPI driver to add stacked and parallel memories support. The first nine patches https://lore.kernel.org/all/20230119185342.2093323-1-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-2-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-3-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-4-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-5-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-6-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-7-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-8-amit.kumar-mahapatra@a… https://lore.kernel.org/all/20230310173217.3429788-9-amit.kumar-mahapatra@a… of the previous series got applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next But the rest of the patches in the series did not get applied as failure was reported for spi driver with GPIO CS, so send the remaining patches in the series after rebasing it on top of for-next branch and fixing the issue. CS GPIO is not tested on our hardware, but it has been tested by @Stefan https://lore.kernel.org/all/005001da1efc$619ad5a0$24d080e0$@opensource.cirr… --- BRANCH: for-next Changes in v11: - Rebased patch series on top of latest for-next branch. - Added a new patch(1/10) to replace spi->chip_select with spi_get_chipselect() call in tps6594-spi.c. - Added a new patch(2/10) to replace spi->chip_select with spi_get_chipseletc() call in cs35l56_hda_spi.c. - In spi.c initialized unused CS[] to 0xff and spi->cs_index_mask to 0x01 in all flows. - Updated spi_dev_check() to compare the CS of old spi device with the new spi device CS. - Updated cover letter description to add information regarding GPIO CS testing and added Stefen's Tested-by tag in 3/10 patch. Changes in v10: - Rebased patch series on top of latest for-next branch and fixed merge conflicts. Changes in v9: - Updated 1/8 patch description to add an high-level overview of parallel(multi-cs) & stacked design. - Initialized all unused CS to 0xFF. - Moved CS check from spi_add_device() to __spi_add_device(). - Updated __spi_add_device() to check to make sure that multiple logical CS don't map to the same physical CS and same physical CS doesn't map to different logical CS. - Updated 1/8, 5/8 & 7/8 to support arbitrary number of flash devices connected in parallel or stacked mode. - Updated documentation for chip_select. - Added a new spi-nor structure member nor->num_flash to keep track of the number of flashes connected. - Added a new patch in the series 4/8 to move write_enable call just before spi_mem ops call in SPI-NOR. - Added comments in SPI core & SPI-NOR. - Rebased the patch series on top of the latest for-next branch. Changes in v8: - Updated __spi_add_device() and spi_set_cs() to fix spi driver failure with GPIO CS. - Rebased the patch series on top of latest for-next branch and fixed merge conflicts. - Updated cover letter description to add information regarding GPIO CS testing and request Stefan to provide his Tested-by tag for 1/7 patch. - Updated 1/7 patch description. Changes in v7: - Updated spi_dev_check() to avoid failures for spi driver GPIO CS and moved the error message from __spi_add_device() to spi_dev_check(). - Resolved code indentation issue in spi_set_cs(). - In spi_set_cs() call spi_delay_exec( ) once if the controller supports multi cs with both the CS backed by GPIO. - Updated __spi_validate()to add checks for both the GPIO CS. - Replaced cs_index_mask bit mask with SPI_CS_CNT_MAX. - Updated struct spi_controller to represent multi CS capability of the spi controller through a flag bit SPI_CONTROLLER_MULTI_CS instead of a boolen structure member "multi_cs_cap". - Updated 1/7 patch description . Changes in v6: - Rebased on top of latest v6.3-rc1 and fixed merge conflicts in spi-mpc512x-psc.c, sfdp.c, spansion.c files and removed spi-omap-100k.c. - Updated spi_dev_check( ) to reject new devices if any one of the chipselect is used by another device. Changes in v5: - Rebased the patches on top of v6.3-rc1 and fixed the merge conflicts. - Fixed compilation warnings in spi-sh-msiof.c with shmobile_defconfig Changes in v4: - Fixed build error in spi-pl022.c file - reported by Mark. - Fixed build error in spi-sn-f-ospi.c file. - Added Reviewed-by: Serge Semin <fancer.lancer(a)gmail.com> tag. - Added two more patches to replace spi->chip_select with API calls in mpc832x_rdb.c & cs35l41_hda_spi.c files. Changes in v3: - Rebased the patches on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi.git for-next - Added a patch to convert spi_nor_otp_region_len(nor) & spi_nor_otp_n_regions(nor) macros into inline functions - Added Reviewed-by & Acked-by tags Changes in v2: - Rebased the patches on top of v6.2-rc1 - Created separate patch to add get & set APIs for spi->chip_select & spi->cs_gpiod, and replaced all spi->chip_select and spi->cs_gpiod references with the API calls. - Created separate patch to add get & set APIs for nor->params. --- Amit Kumar Mahapatra (10): mfd: tps6594: Use set/get APIs to access spi->chip_select ALSA: hda/cs35l56: Use set/get APIs to access spi->chip_select spi: Add multi-cs memories support in SPI core mtd: spi-nor: Convert macros with inline functions mtd: spi-nor: Add APIs to set/get nor->params mtd: spi-nor: Move write enable inside specific write & erase APIs mtd: spi-nor: Add stacked memories support in spi-nor spi: spi-zynqmp-gqspi: Add stacked memories support in GQSPI driver mtd: spi-nor: Add parallel memories support in spi-nor spi: spi-zynqmp-gqspi: Add parallel memories support in GQSPI driver drivers/mfd/tps6594-spi.c | 2 +- drivers/mtd/spi-nor/atmel.c | 17 +- drivers/mtd/spi-nor/core.c | 683 +++++++++++++++++++++++++------ drivers/mtd/spi-nor/core.h | 8 + drivers/mtd/spi-nor/debugfs.c | 4 +- drivers/mtd/spi-nor/gigadevice.c | 4 +- drivers/mtd/spi-nor/issi.c | 11 +- drivers/mtd/spi-nor/macronix.c | 10 +- drivers/mtd/spi-nor/micron-st.c | 35 +- drivers/mtd/spi-nor/otp.c | 48 ++- drivers/mtd/spi-nor/sfdp.c | 33 +- drivers/mtd/spi-nor/spansion.c | 57 +-- drivers/mtd/spi-nor/sst.c | 7 +- drivers/mtd/spi-nor/swp.c | 25 +- drivers/mtd/spi-nor/winbond.c | 2 +- drivers/mtd/spi-nor/xilinx.c | 18 +- drivers/spi/spi-zynqmp-gqspi.c | 58 ++- drivers/spi/spi.c | 259 ++++++++++-- include/linux/mtd/spi-nor.h | 21 +- include/linux/spi/spi.h | 51 ++- sound/pci/hda/cs35l56_hda_spi.c | 2 +- 21 files changed, 1085 insertions(+), 270 deletions(-) -- 2.17.1

10 61

[PATCH v3] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 19 Aug '24

19 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..32954c5a63c3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,26 +22569,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22598,9 +22614,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Port API coverage reporting to Python 3 and adapt to current doxygen output
by GitHub pull_request - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-python pull request #15 was opened from tormodvolden: Tested by comparing the new output from alsa-python-coverage.py with the old COVERAGE file. For transparency and easier review there are multiple commits for the Python 3 fixups, but this can be squashed if desired. Request URL : https://github.com/alsa-project/alsa-python/pull/15 Patch URL : https://github.com/alsa-project/alsa-python/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Add persistent ID's for soundcards and midi devices (like coreaudio)
by GitHub issues - opened 18 Aug '24

18 Aug '24

alsa-project/alsa-lib issue #407 was opened from sjomae: "This is something where macOS' Coreaudio/MIDI shines. Unlike macOS Linux/ALSA has no persistent unique IDs for soundcards or MIDI devices. ALSA supports hotplug, and first come first server sequential numeric IDs. The best you^Wpipewire can do is keep track of cards by name. So this is not something pipewire can reliably address, until ALSA get support to identify cards by vendor and serial-number, and provide a UUID." https://lists.linuxaudio.org/hyperkitty/list/linux-audio-dev@lists.linuxaud… Issue URL : https://github.com/alsa-project/alsa-lib/issues/407 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/1] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml
by Frank Li 17 Aug '24

17 Aug '24

Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Signed-off-by: Frank Li <Frank.Li(a)nxp.com> --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- 2 files changed, 55 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 0000000000000..a42bf9bde6940 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel <sre(a)kernel.org> + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d8..0000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - <int> - I2C slave address - -- Vdd-supply - <phandle> - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; -- 2.34.1

3 2

ASUS Maximus Z790 Hero (ALC4082) - USB Audio 5.1 speakers assigned incorrectly and subwoofer not working
by GitHub issues - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-ucm-conf issue #437 was opened from Gess79: I hope I'm correct to post this here. Still new to things Linux. **Situation** OS: Kubuntu 24.04 Mainboard = ASUS ROG Maximus Z790 Hero USB Audio = 0b05:1a53 ASUSTek Computer, Inc. USB Audio Speakers = Logitech Z906 Surround Kernel = 6.8.0-40-generic Alsa version = 1.2.9 **Issue** The speakers seem to be assigned incorrectly and the subwoofer isn't working. I'm testing the speakers through pulseaudio as follows: ![image](https://github.com/user-attachments/assets/8bf65600-63ad-46cc-af68-d51ab6d4a7e6) I'm listing the results below in three categories: - The icon clicked within pulseaudio - The soundfile played - The speaker the sound is actually output to _Icon clicked - Soundfile played - Physical speaker output_ Front Left - Front Left - Front Left Front Center - Front Center - Rear Left Front Right - Front Right - Front Right Rear Left - Rear Left - Front Center Subwoofer - Rear Center - Rear Right Rear Right - (nothing) - (nothing) I've had a lengthy discussion on Reddit but I couldn't find a fix for it so far: https://www.reddit.com/r/linux4noobs/comments/1980uv1/comment/kieqsor/ I'm happy to provide all information necessary to pinpoint the issue and fix it. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/437 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 - I haven't changed the reference counting, which seems wrong to me in the original code as well. This would be a different issue. Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability - the third commit is not for fixing the crash, but for fixing set_tuple() to actually do anything for IEC958 Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> we seems to have ended up with duplicate clocks for frame-sync on sm8250, it has both va and fsgen which are exactly same things. Remove the redundant va clock and make it align with other SoCs. Codec driver does not even handle va clock, so remove this from the bindings and examples to avoid any confusion. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 ++----------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 06b5f7be3608..6f5644a89feb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -64,6 +64,7 @@ allOf: compatible: enum: - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro - qcom,sm8450-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro then: @@ -79,24 +80,6 @@ allOf: - const: dcodec - const: fsgen - - if: - properties: - compatible: - enum: - - qcom,sm8250-lpass-wsa-macro - then: - properties: - clocks: - minItems: 6 - clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: va - - const: fsgen - - if: properties: compatible: @@ -130,8 +113,7 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, - <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; clock-output-names = "mclk"; }; -- 2.25.1

3 2

[PATCH v1] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 16 Aug '24

16 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781, remove unnecessary line feed, strscpy replace scnprintf. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Use strscpy to replace scnprintf. - rename dai_driver name from "tas2781_codec" to "tasdev_codec", in case of misunderstanding whether tas2781 or tas2563. - Remove unnecessary line feed for tasdevice_dsp_create_ctrls --- sound/soc/codecs/tas2781-i2c.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..6631254d4985 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -353,7 +353,7 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -424,8 +424,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; @@ -453,8 +452,8 @@ static int tasdevice_dsp_create_ctrls( goto out; } - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -462,8 +461,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -673,7 +671,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

4 6

[PATCH] ASoC: SOF: amd: Fix for acp init sequence
by Vijendar Mukunda 16 Aug '24

16 Aug '24

When ACP is not powered on by default, acp power on sequence explicitly invoked by programming pgfsm control mask. The existing implementation checks the same PGFSM status mask and programs the same PGFSM control mask in all ACP variants which breaks acp power on sequence for ACP6.0 and ACP6.3 variants. So to fix this issue, update ACP pgfsm control mask and status mask based on acp descriptor rev field, which will vary based on acp variant. Fixes: 846aef1d7cc0 ("ASoC: SOF: amd: Add Renoir ACP HW support") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 19 +++++++++++++++++-- sound/soc/sof/amd/acp.h | 7 +++++-- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 35e56d22930f..0f6115c8b005 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -433,6 +433,7 @@ static int acp_power_on(struct snd_sof_dev *sdev) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int base = desc->pgfsm_base; unsigned int val; + unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; int ret; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET); @@ -440,9 +441,23 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - if (val & ACP_PGFSM_STATUS_MASK) + switch (desc->rev) { + case 3: + case 5: + acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; + break; + case 6: + acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; + break; + default: + return -EINVAL; + } + + if (val & acp_pgfsm_status_mask) snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + PGFSM_CONTROL_OFFSET, - ACP_PGFSM_CNTL_POWER_ON_MASK); + acp_pgfsm_cntl_mask); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, base + PGFSM_STATUS_OFFSET, val, !val, ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index f6f0fcfeb691..321c40cc6d50 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -25,8 +25,11 @@ #define ACP_REG_POLL_TIMEOUT_US 2000 #define ACP_DMA_COMPLETE_TIMEOUT_US 5000 -#define ACP_PGFSM_CNTL_POWER_ON_MASK 0x01 -#define ACP_PGFSM_STATUS_MASK 0x03 +#define ACP3X_PGFSM_CNTL_POWER_ON_MASK 0x01 +#define ACP3X_PGFSM_STATUS_MASK 0x03 +#define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 +#define ACP6X_PGFSM_STATUS_MASK 0x0F + #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 #define ACP_RELEASE_RESET 0x00 -- 2.34.1

2 1

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - edited 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was edited from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. I don't know if this is expected or if a warning should be issued. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

Use PyBytes and not Unicode strings for IEC958 values
by GitHub pull_request - opened 16 Aug '24

16 Aug '24

alsa-project/alsa-python pull request #14 was opened from tormodvolden: Suggested fix for #13. - the first commit is not strictly related but also needed to avoid hctltest1.py crashing with my audio devices, where some have info.count = 0. Let me know if more information is needed. - I went for PyBytes instead of PyByteArray - for setting values, if the provided PyBytes value is shorter than the value, it will be zero-padded. - if it is too long, it will be silently truncated - following the existing code style, no empty lines were inserted, but this could help readability Request URL : https://github.com/alsa-project/alsa-python/pull/14 Patch URL : https://github.com/alsa-project/alsa-python/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 16 Aug '24

16 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. Add the version info into of_data, so that it does not need to use core_id registers to get version number. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

3 4

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by zhangyi＠everest-semi.com 16 Aug '24

16 Aug '24

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

2 1

[PATCH] ALSA: aoa: Use helper function for_each_child_of_node()
by Zhang Zekun 16 Aug '24

16 Aug '24

for_each_child_of_node can help to iterate through the device_node, and we don't need to use while loop. No functional change with this conversion. Signed-off-by: Zhang Zekun <zhangzekun11(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 5431d2c49421..ce84288168e4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -335,7 +335,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) { - struct device_node *np = NULL; + struct device_node *np; int got = 0, err; struct i2sbus_control *control = NULL; @@ -347,7 +347,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + for_each_child_of_node(dev->ofdev.dev.of_node, np) { if (of_device_is_compatible(np, "i2sbus") || of_device_is_compatible(np, "i2s-modem")) { got += i2sbus_add_dev(dev, control, np); -- 2.17.1

2 1

[PATCH v2 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 15 Aug '24

15 Aug '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v2: - used dev_dbg instead of dev_info - Link to v1: https://lore.kernel.org/r/20240626-port-map-v1-0-bd8987d2b332@linaro.org --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

5 10

[PATCH 0/6] soc: qcom: use scoped device node handling to simplify error paths
by Krzysztof Kozlowski 15 Aug '24

15 Aug '24

Use cleanup.h/scope on device nodes to get rid of of_node_put(). Code is a bit simpler. Best regards, Krzysztof --- Krzysztof Kozlowski (6): soc: qcom: apr: simplify with scoped for each OF child loop soc: qcom: aoss: simplify with scoped for each OF child loop soc: qcom: ice: use scoped device node handling to simplify error paths soc: qcom: ocmem: use scoped device node handling to simplify error paths soc: qcom: pbs: use scoped device node handling to simplify error paths soc: qcom: smp2p: use scoped device node handling to simplify error paths drivers/soc/qcom/apr.c | 5 +---- drivers/soc/qcom/ice.c | 14 +++++--------- drivers/soc/qcom/ocmem.c | 7 ++----- drivers/soc/qcom/qcom-pbs.c | 16 ++++++---------- drivers/soc/qcom/qcom_aoss.c | 8 +++----- drivers/soc/qcom/smp2p.c | 16 ++++------------ 6 files changed, 21 insertions(+), 45 deletions(-) --- base-commit: d54f33dafd9e93367ffaf95318ed371138a6c8c0 change-id: 20240813-b4-cleanup-h-of-node-put-other-7e9c7128abe4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 7

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - edited 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was edited from tormodvolden: Version 1.2.12, on Ubuntu 24.04, Python 3.12.3. It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - edited 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was edited from tormodvolden: Version 1.2.12, on Ubuntu 24.04, Python 3.12. It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

hctltest1.py crashes with utf-8 decoding error on IEC958 "status"
by GitHub issues - opened 15 Aug '24

15 Aug '24

alsa-project/alsa-python issue #13 was opened from tormodvolden: It seems like the "status" member returned by `snd_ctl_elem_value_get_iec958(self->value, iec958);` on pyalsa/alsahcontrol.c line 1053 starts with 0xff then a 0x00. I haven't been able to find a corresponding node under /proc/asound to check this. The relevant aplay -L entry is ``` iec958:CARD=Stereo,DEV=0 Logitech H570e Stereo, USB Audio IEC958 (S/PDIF) Digital Audio Output ``` Here is a dump of /proc/asound/card1 (card 1 is called "Stereo") <details> ``` $ more $(find /proc/asound/card1 -type f) :::::::::::::: /proc/asound/card1/id :::::::::::::: Stereo :::::::::::::: /proc/asound/card1/pcm0c/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: CAPTURE id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 :::::::::::::: /proc/asound/card1/pcm0c/sub0/status :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/hw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0c/sub0/sw_params :::::::::::::: closed :::::::::::::: /proc/asound/card1/pcm0p/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/info :::::::::::::: card: 1 device: 0 subdevice: 0 stream: PLAYBACK id: USB Audio name: USB Audio subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 :::::::::::::: /proc/asound/card1/pcm0p/sub0/status :::::::::::::: state: RUNNING owner_pid : 1928 trigger_time: 270044.979966617 tstamp : 309371.219688371 delay : 1872 avail : 31520 avail_max : 32032 ----- hw_ptr : 1887652896 appl_ptr : 1887654144 :::::::::::::: /proc/asound/card1/pcm0p/sub0/hw_params :::::::::::::: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 256 buffer_size: 32768 :::::::::::::: /proc/asound/card1/pcm0p/sub0/sw_params :::::::::::::: tstamp_mode: ENABLE period_step: 1 avail_min: 256 start_threshold: 9223372036854775807 stop_threshold: 32768 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 :::::::::::::: /proc/asound/card1/usbid :::::::::::::: 046d:0a56 :::::::::::::: /proc/asound/card1/usbbus :::::::::::::: 003/006 :::::::::::::: /proc/asound/card1/stream0 :::::::::::::: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed : USB A udio Playback: Status: Running Interface = 2 Altset = 1 Packet Size = 192 Momentary freq = 48000 Hz (0x30.0000) Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x01 (1 OUT) (ADAPTIVE) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 2 Endpoint: 0x81 (1 IN) (ASYNC) Rates: 8000, 16000, 24000, 32000, 44100, 48000 Bits: 16 Channel map: FL FR :::::::::::::: /proc/asound/card1/usbmixer :::::::::::::: USB Mixer: usb_id=0x046d0a56, ctrlif=0, ctlerr=0 Card: Logitech Inc Logitech H570e Stereo at usb-0000:00:14.0-4.4.2, full speed Unit: 2 Control: name="Auto Gain Control", index=0 Info: id=2, control=7, cmask=0x0, channels=1, type="BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 2 Control: name="Headset Capture Volume", index=0 Info: id=2, control=2, cmask=0x0, channels=1, type="S16" Volume: min=-7680, max=1280, dBmin=-3000, dBmax=500 Unit: 2 Control: name="Headset Capture Switch", index=0 Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 5 Control: name="Headset Playback Volume", index=0 Info: id=5, control=2, cmask=0x3, channels=2, type="S16" Volume: min=-15360, max=1280, dBmin=-6000, dBmax=500 Unit: 5 Control: name="Headset Playback Switch", index=0 Info: id=5, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 Unit: 11 Control: name="Sidetone Playback Volume", index=0 Info: id=11, control=2, cmask=0x0, channels=1, type="S16" Volume: min=0, max=4125, dBmin=0, dBmax=1611 Unit: 11 Control: name="Sidetone Playback Switch", index=0 Info: id=11, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 ``` </details> This is the last part of the output from running `python3 test/hctltest1.py ` ``` ***** count: 1 device: 0 dimensions: None element: <alsahcontrol.Element object at 0x7e418ebb9650> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 is_inactive: False is_locked: False is_owner: False is_readable: True is_tlv_Commandable: False is_tlv_Writable: False is_tlv_readable: False is_user: False is_volatile: False is_writable: False name: IEC958 Playback Con Mask numid: 19 owner: -1 subdevice: 0 type: 5 (IEC958) ----- device: 0 element: <alsahcontrol.Element object at 0x7e418ebb9650> get_array: <built-in method get_array of alsahcontrol.Value object at 0x7e418ebb9620> get_tuple: <built-in method get_tuple of alsahcontrol.Value object at 0x7e418ebb9620> id: (19, 2, 0, 0, 'IEC958 Playback Con Mask', 0) index: 0 interface: 2 name: IEC958 Playback Con Mask numid: 19 read: <built-in method read of alsahcontrol.Value object at 0x7e418ebb9620> set_array: <built-in method set_array of alsahcontrol.Value object at 0x7e418ebb9620> set_tuple: <built-in method set_tuple of alsahcontrol.Value object at 0x7e418ebb9620> subdevice: 0 write: <built-in method write of alsahcontrol.Value object at 0x7e418ebb9620> UnicodeDecodeError: 'utf-8' codec can't decode byte 0xff in position 1: invalid start byte The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 69, in <module> value(element1) File "/home/tormod/SOURCES/pyalsa-debian/python-pyalsa/test/hctltest1.py", line 36, in value values = value.get_tuple(info.type, info.count) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <method 'get_tuple' of 'alsahcontrol.Value' objects> returned a result with an exception set ``` info.type is 5 info.count is 1 Issue URL : https://github.com/alsa-project/alsa-python/issues/13 Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ASoc: tas2781: Remove unnecessary line feed and space
by Shenghao Ding 15 Aug '24

15 Aug '24

Remove unnecessary line feed for tasdevice_dsp_create_ctrls, and remove two unnecessary spaces in tas2563_digital_gain_get and tas2563_digital_gain_put. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8c9efd9183ff..9a32e0504857 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -157,7 +157,7 @@ static int tas2563_digital_gain_get( mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); goto out; @@ -203,7 +203,7 @@ static int tas2563_digital_gain_put( vol = clamp(vol, 0, max); mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); rc = -1; @@ -423,8 +423,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; -- 2.34.1

2 1

[PATCH alsa-lib] test: Add an example program to inquire UMP Endpoint and Block info
by Takashi Iwai 15 Aug '24

15 Aug '24

Provide an example program to inquire UMP Endpoint and Block info in various APIs. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- test/Makefile.am | 3 +- test/umpinfo.c | 147 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 149 insertions(+), 1 deletion(-) create mode 100644 test/umpinfo.c diff --git a/test/Makefile.am b/test/Makefile.am index 635fa39bb7e3..707063cc9fc0 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -1,7 +1,7 @@ SUBDIRS=. lsb check_PROGRAMS=control pcm pcm_min latency seq seq-ump-example \ - playmidi1 timer rawmidi midiloop \ + playmidi1 timer rawmidi midiloop umpinfo \ oldapi queue_timer namehint client_event_filter \ chmap audio_time user-ctl-element-set pcm-multi-thread @@ -17,6 +17,7 @@ playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la rawmidi_LDADD=../src/libasound.la midiloop_LDADD=../src/libasound.la +umpinfo_LDADD=../src/libasound.la oldapi_LDADD=../src/libasound.la queue_timer_LDADD=../src/libasound.la namehint_LDADD=../src/libasound.la diff --git a/test/umpinfo.c b/test/umpinfo.c new file mode 100644 index 000000000000..92a209361861 --- /dev/null +++ b/test/umpinfo.c @@ -0,0 +1,147 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// Inquire UMP Endpoint and Block info in various APIs + +#include <stdio.h> +#include <alsa/asoundlib.h> + +enum { UNKNOWN, SEQ, RAWMIDI, CTL }; +static int mode = UNKNOWN; + +static snd_seq_t *seq; +static int client; +static snd_ump_t *ump; +static snd_ctl_t *ctl; + +static int get_ump_endpoint_info(snd_ump_endpoint_info_t *ep) +{ + switch (mode) { + case SEQ: + return snd_seq_get_ump_endpoint_info(seq, client, ep); + case RAWMIDI: + return snd_ump_endpoint_info(ump, ep); + case CTL: + return snd_ctl_ump_endpoint_info(ctl, ep); + default: + return -EINVAL; + } +} + +static int get_ump_block_info(int blk, snd_ump_block_info_t *bp) +{ + switch (mode) { + case SEQ: + return snd_seq_get_ump_block_info(seq, client, blk, bp); + case RAWMIDI: + snd_ump_block_info_set_block_id(bp, blk); + return snd_ump_block_info(ump, bp); + case CTL: + snd_ump_block_info_set_block_id(bp, blk); + return snd_ctl_ump_block_info(ctl, bp); + default: + return -EINVAL; + } +} + +static void help(void) +{ + printf("umpinfo: inquire UMP Endpoint and Block info\n" + " umpinfo -c target inquiry via control API\n" + " umpinfo -m target inquiry via UMP rawmidi API\n" + " umpinfo -s client inquiry via sequencer API\n" + " target = hw:0, etc\n" + " client = sequencer client number\n"); + exit(1); +} + +int main(int argc, char **argv) +{ + const char *target = "hw:0"; + snd_ump_endpoint_info_t *ep; + snd_ump_block_info_t *blk; + const unsigned char *s; + int c, i, num_blks; + + while ((c = getopt(argc, argv, "s:m:c:h")) >= 0) { + switch (c) { + case 's': + mode = SEQ; + client = atoi(target); + break; + case 'm': + mode = RAWMIDI; + target = optarg; + break; + case 'c': + mode = CTL; + target = optarg; + break; + default: + help(); + break; + } + } + + switch (mode) { + case SEQ: + if (snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0)) { + fprintf(stderr, "failed to open sequencer\n"); + return 1; + } + break; + case RAWMIDI: + if (snd_ump_open(&ump, NULL, target, 0)) { + fprintf(stderr, "failed to open UMP rawmidi\n"); + return 1; + } + break; + case CTL: + if (snd_ctl_open(&ctl, target, 0)) { + fprintf(stderr, "failed to open control\n"); + return 1; + } + break; + default: + help(); + break; + } + + snd_ump_endpoint_info_alloca(&ep); + snd_ump_block_info_alloca(&blk); + + if (get_ump_endpoint_info(ep)) { + fprintf(stderr, "failed to get UMP EP info\n"); + return 1; + } + + printf("Name: %s\n", snd_ump_endpoint_info_get_name(ep)); + printf("Product ID: %s\n", snd_ump_endpoint_info_get_product_id(ep)); + printf("Flags: 0x%x\n", snd_ump_endpoint_info_get_flags(ep)); + printf("Protocol Caps: 0x%x\n", snd_ump_endpoint_info_get_protocol_caps(ep)); + printf("Protocol: 0x%x\n", snd_ump_endpoint_info_get_protocol(ep)); + printf("Num Blocks: %d\n", snd_ump_endpoint_info_get_num_blocks(ep)); + printf("Version: 0x%x\n", snd_ump_endpoint_info_get_version(ep)); + printf("Manufacturer ID: 0x%x\n", snd_ump_endpoint_info_get_manufacturer_id(ep)); + printf("Family ID: 0x%x\n", snd_ump_endpoint_info_get_family_id(ep)); + printf("Model ID: 0x%x\n", snd_ump_endpoint_info_get_model_id(ep)); + s = snd_ump_endpoint_info_get_sw_revision(ep); + printf("SW Revision: %02x:%02x:%02x:%02x\n", s[0], s[1], s[2], s[3]); + num_blks = snd_ump_endpoint_info_get_num_blocks(ep); + for (i = 0; i < num_blks; i++) { + if (get_ump_block_info(i, blk)) { + fprintf(stderr, "failed to get UMP Block info for %d\n", i); + continue; + } + printf("\n"); + printf("Block ID: %d\n", snd_ump_block_info_get_block_id(blk)); + printf("Name: %s\n", snd_ump_block_info_get_name(blk)); + printf("Active: %d\n", snd_ump_block_info_get_active(blk)); + printf("Flags: 0x%x\n", snd_ump_block_info_get_flags(blk)); + printf("Direction: %d\n", snd_ump_block_info_get_direction(blk)); + printf("First Group: %d\n", snd_ump_block_info_get_first_group(blk)); + printf("Num Groups: %d\n", snd_ump_block_info_get_num_groups(blk)); + printf("MIDI-CI Version: %d\n", snd_ump_block_info_get_midi_ci_version(blk)); + printf("Sysex8 Streams: %d\n", snd_ump_block_info_get_sysex8_streams(blk)); + } + + return 0; +} -- 2.43.0

1 0

[PATCH alsa-lib] control: Make ump_{endpoint|block}_info calls optional
by Takashi Iwai 15 Aug '24

15 Aug '24

Add the NULL check for ump_endpoint_info and ump_block_info calls. Those can be NULl depending on the target. Fixes: 81b0cf46d16a ("control: Add UMP Endpoint and Block info query support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/control/control.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index d77ab24cc69c..e443d5436103 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -1293,7 +1293,10 @@ int snd_ctl_ump_next_device(snd_ctl_t *ctl, int *device) int snd_ctl_ump_endpoint_info(snd_ctl_t *ctl, snd_ump_endpoint_info_t *info) { assert(ctl && info); - return ctl->ops->ump_endpoint_info(ctl, info); + fprintf(stderr, "%s:%d\n", __func__, __LINE__); + if (ctl->ops->ump_endpoint_info) + return ctl->ops->ump_endpoint_info(ctl, info); + return -ENXIO; } /** @@ -1305,7 +1308,9 @@ int snd_ctl_ump_endpoint_info(snd_ctl_t *ctl, snd_ump_endpoint_info_t *info) int snd_ctl_ump_block_info(snd_ctl_t *ctl, snd_ump_block_info_t *info) { assert(ctl && info); - return ctl->ops->ump_block_info(ctl, info); + if (ctl->ops->ump_block_info) + return ctl->ops->ump_block_info(ctl, info); + return -ENXIO; } /** -- 2.43.0

1 0

[PATCH 5.10 031/352] ARM: pxa: spitz: use gpio descriptors for audio
by Greg Kroah-Hartman 15 Aug '24

15 Aug '24

5.10-stable review patch. If anyone has any objections, please let me know. ------------------ From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 726d8c965bae2d7468445d990849e281dca8cbf7 ] The audio driver should not use a hardwired gpio number from the header. Change it to use a lookup table. Acked-by: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Stable-dep-of: 78ab3d352f29 ("ARM: spitz: fix GPIO assignment for backlight") Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- arch/arm/mach-pxa/spitz.c | 33 ++++++++++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 2 +- arch/arm/mach-pxa/spitz_pm.c | 2 +- sound/soc/pxa/spitz.c | 58 ++++++++------------ 4 files changed, 57 insertions(+), 38 deletions(-) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (99%) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 264de0bc97d68..8f880dbc05fdb 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -43,7 +43,7 @@ #include <linux/platform_data/mmc-pxamci.h> #include <linux/platform_data/usb-ohci-pxa27x.h> #include <linux/platform_data/video-pxafb.h> -#include <mach/spitz.h> +#include "spitz.h" #include "sharpsl_pm.h" #include <mach/smemc.h> @@ -954,11 +954,42 @@ static void __init spitz_i2c_init(void) static inline void spitz_i2c_init(void) {} #endif +static struct gpiod_lookup_table spitz_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.1", SPITZ_GPIO_MIC_BIAS - SPITZ_SCP2_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct gpiod_lookup_table akita_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("i2c-max7310", AKITA_GPIO_MIC_BIAS - AKITA_IOEXP_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + /****************************************************************************** * Audio devices ******************************************************************************/ static inline void spitz_audio_init(void) { + if (machine_is_akita()) + gpiod_add_lookup_table(&akita_audio_gpio_table); + else + gpiod_add_lookup_table(&spitz_audio_gpio_table); + platform_device_register_simple("spitz-audio", -1, NULL, 0); } diff --git a/arch/arm/mach-pxa/include/mach/spitz.h b/arch/arm/mach-pxa/spitz.h similarity index 99% rename from arch/arm/mach-pxa/include/mach/spitz.h rename to arch/arm/mach-pxa/spitz.h index 04828d8918aa3..f97e3ebd762d5 100644 --- a/arch/arm/mach-pxa/include/mach/spitz.h +++ b/arch/arm/mach-pxa/spitz.h @@ -11,7 +11,7 @@ #define __ASM_ARCH_SPITZ_H 1 #endif -#include "irqs.h" /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ +#include <mach/irqs.h> /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ #include <linux/fb.h> /* Spitz/Akita GPIOs */ diff --git a/arch/arm/mach-pxa/spitz_pm.c b/arch/arm/mach-pxa/spitz_pm.c index 25a1f8c5a7382..6167f96d7b41e 100644 --- a/arch/arm/mach-pxa/spitz_pm.c +++ b/arch/arm/mach-pxa/spitz_pm.c @@ -20,7 +20,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> -#include <mach/spitz.h> +#include "spitz.h" #include "pxa27x.h" #include "sharpsl_pm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 7c1384a869ca4..44303b6eb228b 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -14,13 +14,12 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-i2s.h" @@ -37,7 +36,7 @@ static int spitz_jack_func; static int spitz_spk_func; -static int spitz_mic_gpio; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { @@ -56,8 +55,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 1); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -65,8 +64,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -74,8 +73,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -83,8 +82,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_HP_OFF: @@ -93,8 +92,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; } @@ -199,7 +198,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -287,39 +286,28 @@ static int spitz_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (machine_is_akita()) - spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; - else - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - - ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); - if (ret) - goto err1; - - ret = gpio_direction_output(spitz_mic_gpio, 0); - if (ret) - goto err2; + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err2; - } - - return 0; -err2: - gpio_free(spitz_mic_gpio); -err1: return ret; } static int spitz_remove(struct platform_device *pdev) { - gpio_free(spitz_mic_gpio); return 0; } -- 2.43.0

1 0

[PATCH 5.15 039/484] ARM: pxa: spitz: use gpio descriptors for audio
by Greg Kroah-Hartman 15 Aug '24

15 Aug '24

5.15-stable review patch. If anyone has any objections, please let me know. ------------------ From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit 726d8c965bae2d7468445d990849e281dca8cbf7 ] The audio driver should not use a hardwired gpio number from the header. Change it to use a lookup table. Acked-by: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Reviewed-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Stable-dep-of: 78ab3d352f29 ("ARM: spitz: fix GPIO assignment for backlight") Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- arch/arm/mach-pxa/spitz.c | 33 ++++++++++- arch/arm/mach-pxa/{include/mach => }/spitz.h | 2 +- arch/arm/mach-pxa/spitz_pm.c | 2 +- sound/soc/pxa/spitz.c | 58 ++++++++------------ 4 files changed, 57 insertions(+), 38 deletions(-) rename arch/arm/mach-pxa/{include/mach => }/spitz.h (99%) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 264de0bc97d68..8f880dbc05fdb 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -43,7 +43,7 @@ #include <linux/platform_data/mmc-pxamci.h> #include <linux/platform_data/usb-ohci-pxa27x.h> #include <linux/platform_data/video-pxafb.h> -#include <mach/spitz.h> +#include "spitz.h" #include "sharpsl_pm.h" #include <mach/smemc.h> @@ -954,11 +954,42 @@ static void __init spitz_i2c_init(void) static inline void spitz_i2c_init(void) {} #endif +static struct gpiod_lookup_table spitz_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.1", SPITZ_GPIO_MIC_BIAS - SPITZ_SCP2_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct gpiod_lookup_table akita_audio_gpio_table = { + .dev_id = "spitz-audio", + .table = { + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_L - SPITZ_SCP_GPIO_BASE, + "mute-l", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("sharp-scoop.0", SPITZ_GPIO_MUTE_R - SPITZ_SCP_GPIO_BASE, + "mute-r", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("i2c-max7310", AKITA_GPIO_MIC_BIAS - AKITA_IOEXP_GPIO_BASE, + "mic", GPIO_ACTIVE_HIGH), + { }, + }, +}; + /****************************************************************************** * Audio devices ******************************************************************************/ static inline void spitz_audio_init(void) { + if (machine_is_akita()) + gpiod_add_lookup_table(&akita_audio_gpio_table); + else + gpiod_add_lookup_table(&spitz_audio_gpio_table); + platform_device_register_simple("spitz-audio", -1, NULL, 0); } diff --git a/arch/arm/mach-pxa/include/mach/spitz.h b/arch/arm/mach-pxa/spitz.h similarity index 99% rename from arch/arm/mach-pxa/include/mach/spitz.h rename to arch/arm/mach-pxa/spitz.h index 04828d8918aa3..f97e3ebd762d5 100644 --- a/arch/arm/mach-pxa/include/mach/spitz.h +++ b/arch/arm/mach-pxa/spitz.h @@ -11,7 +11,7 @@ #define __ASM_ARCH_SPITZ_H 1 #endif -#include "irqs.h" /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ +#include <mach/irqs.h> /* PXA_NR_BUILTIN_GPIO, PXA_GPIO_TO_IRQ */ #include <linux/fb.h> /* Spitz/Akita GPIOs */ diff --git a/arch/arm/mach-pxa/spitz_pm.c b/arch/arm/mach-pxa/spitz_pm.c index 25a1f8c5a7382..6167f96d7b41e 100644 --- a/arch/arm/mach-pxa/spitz_pm.c +++ b/arch/arm/mach-pxa/spitz_pm.c @@ -20,7 +20,7 @@ #include <asm/mach-types.h> #include <mach/hardware.h> -#include <mach/spitz.h> +#include "spitz.h" #include "pxa27x.h" #include "sharpsl_pm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 7c1384a869ca4..44303b6eb228b 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -14,13 +14,12 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-i2s.h" @@ -37,7 +36,7 @@ static int spitz_jack_func; static int spitz_spk_func; -static int spitz_mic_gpio; +static struct gpio_desc *gpiod_mic, *gpiod_mute_l, *gpiod_mute_r; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { @@ -56,8 +55,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 1); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 1); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -65,8 +64,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -74,8 +73,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -83,8 +82,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 1); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 1); break; case SPITZ_HP_OFF: @@ -93,8 +92,8 @@ static void spitz_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); - gpio_set_value(SPITZ_GPIO_MUTE_L, 0); - gpio_set_value(SPITZ_GPIO_MUTE_R, 0); + gpiod_set_value(gpiod_mute_l, 0); + gpiod_set_value(gpiod_mute_r, 0); break; } @@ -199,7 +198,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); + gpiod_set_value_cansleep(gpiod_mic, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -287,39 +286,28 @@ static int spitz_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (machine_is_akita()) - spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; - else - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - - ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); - if (ret) - goto err1; - - ret = gpio_direction_output(spitz_mic_gpio, 0); - if (ret) - goto err2; + gpiod_mic = devm_gpiod_get(&pdev->dev, "mic", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mic)) + return PTR_ERR(gpiod_mic); + gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_l)) + return PTR_ERR(gpiod_mute_l); + gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_mute_r)) + return PTR_ERR(gpiod_mute_r); card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err2; - } - - return 0; -err2: - gpio_free(spitz_mic_gpio); -err1: return ret; } static int spitz_remove(struct platform_device *pdev) { - gpio_free(spitz_mic_gpio); return 0; } -- 2.43.0

1 0

[PATCH v11] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by Baojun Xu 15 Aug '24

15 Aug '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. The code support Realtek as the primary codec. In patch version-10, add multi devices firmware binary support, to compatble with windows driver, they can share same firmware binary. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v11: - Define config number for pre-reset and post-reset scripts. - Remove acpi_subsystem_id as it was not used. - Change variables order in structure, to avoid possible memory holes. - Remove pre-reset and post-reset scripts in reset function. - Change variables name to more common. - Add pre-reset and post-reset scripts in fw_ready. - Add empty line after end of function. - Remove reset function before fw ready. - Add result check for tasdevice_spi_prmg_load(). - Return real result of tasdevice_load_block_kernel(). - Change code for ppcver check, and update debug message. - Remove load_calib_data() as no calibration data in firmware now. - Remove tasdevice_spi_select_tuningprm_cfg() as config will be forced downloaded after program download. - Change comments for tasdevice_spi_tuning_switch(). - Remove unnecessary brace. v10: - Remove project id 103c8b93 in Patch_realtek.c, as it was used for test only, offcial project id is 103c8daa (Gemtree). - Add TAS2781_REG_CLK_CONFIG for check if AMP has been reset. - Add table for pre-reset and post-reset. - Perform pre-reset config before reset AMP. - Perform post-reset config after reset AMP. - Fix wrong offset in calibration data issue. - Remove tuning switch after firmware download. - Force stop play in runtime suspend. - Apply patch for component change in bind & unbind. - Remove program download and calibration data apply, switch tuning only in runtime resume. - Check if AMP has reseted in system resume, and do firmware download and calibration data apply. - Change flag 0xCx to 0x8x for coeff data force download. - Add device index for multi device firmware binary file compatible. - Check device index before registers write during firmware download. - Force config data download after program download. - Remove config download in tuning switch. v9: - Add parentheses for variables in micro define. - Add define for calibration registers address identify. - Change variable name from calibration_data to cali_data. - Add variable for calibration registers address in struct tasdevice_priv. - Change format for alignment to parentheses. - Remove tas2781_reg_defaults as it will be over write in firmware download. - Remove RESET for second device in tas2781_spi_reset(), as it will cause no sound on second device. - Add calibration registers address initial in tasdevice_spi_init(). - Remove unnecessary parentheses. - Remove exception in tas2781_hda_playback_hook(). - Remove hard code for calibration registers address. - Add support for calibration data format V3 for registers address identify, as new firmware tool will define calibration registers address dynamic, so can't hard code it, need read it from BIOS with calibration data. - Remove some debug print. - Change hard code value (3) to ARRAY_SIZE(tas2781_snd_controls)/2. - New patch apply for change position of component_del(). - Change label from out, out1 to config_err and block_err. - Remove unnecessary parentheses in tas2781_spi_fwlib.c. - Remove tasdevice_spi_prmg_calibdata_load(). v8: - Move *-y position in sound/pci/hda/Makefile. - Change + to | for bit operation in macro define. - Change TASDEVICE_Checksum to TASDEVICE_CHECKSUM. v7: - Add new project Gemtree (0x103c8daa) support for tas2781 spi driver. v6: - Remove client in tasdevice_priv struct as it was not used. - Change format for tasdevice_spi_dev_write() in tas2781_hda_spi.c. - Change update bits function to read & write mode, as tas2781 use last bit to mask read & write, it cause regmap_update_bits work not as our expected. - Add element in tas2781_snd_controls, tas2781_prof_ctrl, tas2781_dsp_conf_ctrl, to support second device. Previous version add sound card for first index only. - Change calibration registers address declaire to TASDEVICE_REG() mode. - Add snd_ctl_new1() for second device. - Change format for tas2781_hda_unbind() to fit 79 chars in one line. - Remove client for spi as it was not used. - Remove unused variables. - Remove cur_prog clean in runtime_suspend(). - Remove unused variables. - Add tasdevice_spi_tuning_switch() in runtime_resume. - Remove cur_prog clean in system_resume(). - Add tuning switch if playback started in system_resume(). - Change TASDEVICE_PAGE_REG to TASDEVICE_REG_ID in tas2781_spi_fwlib.c. - Remove calibration binary load functions as data will be loaded from EFI. v5: - Combined all modification into one patch for pass the compile. - Remove old hardware id "TIAS2781" as no production on market. - Change max page size to 256 in tas2781-spi.h. - Change book reg from 127 to GENMASK(7, 1), as one bit shift is needed. - Change register address define as one bit shift needed for offset. - Change block check sum comments to Firmware block from I2C. - Change define of (page + reg) to TASDEVICE_PAGE_REG from TASDEVICE_PGRG. - Change to lower case in defined value. - Add registers default value table in tas2781_hda_spi.c - Change rang_max to 256 * 256. - Add zero_flag_mask in regmap_config. - Change max_register to 256 * 256. - Add registers default table into structure regmap_config. - Change parameter from book number to whole register address. - Remove page 0 switch as SPI driver will do it. - Add bit(0) for read function as tas2781 will check last bit of address, to determine operation mode is read (1) or write (0). - Change switch case to if check in tas2781_hda_playback_hook(), as it may cause compile problem in clang-19 building. - Change variable declaration position. - Change order in if check. - Remove old hardware id ("TIAS2781") support. - Remove debug message in runtime suspend and resume. - Remove regmap.h include in tas2781_spi_fwlib.c. - Remove MODULE_ in fwlib as this file will work with tas2781_hda_spi.c only. v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs - Move include of sound/tas2781-dsp.h to c source file tas2781_hda_spi.c, tas2781_spi_fwlib.c, as it's not needed in header file tas2781-spi.h - Use page and book number define in other micro define - Change 4000000 to (4 * HZ_PER_MHZ) - Add define for calibration variable name and size in UEFI. - Remove structure tasdevice & tasdevice_irqinfo, and move variables in to structure tasdevice_priv - Remove some unused variables in sttructure tasdevice_priv - Remove function declare of tascodec_init(), use tascodec_spi_init() - Remove function declare of tasdevice_remove() - Add array_size.h included in tas2781_hda_spi.c - Add cleanup.h included for change mutex to guard - Add units.h included - Include tas2781-dsp.h - Remove unused variables in tas2781_hda structure - Change 0xff to GENMASK(7, 0) - Change 128 to define of max page size - Change code as all variables was moved to tasdevice_priv structure - Change comments for error in book write, or page write - Remove initial of ret, and return 0 directly - Change comments for wrong typo, add space in front and end - Add check for bulk read, max size should be a page size - Change usleep_rang() to fsleep() - Change mutex_lock to guard(mutex), and remove mutex_unlock() - Change tasdevice_spi_init() to void from int as no value return required - Change second parameter to sizeof - Remove tasdevice_clamp(), use clamp() - Add compatible with old hardware id "TIAS2781" - Remove cs_gpio and spi device in tas2781_read_acpi() as use default CS - Change dev-index check, return -EINVAL if it's <= 0 - Change 0xFF to U8_MAX - Remove GPIO get for chip-select, use default from SPI device - Perform RESET in first AMP only as all AMPs share same RESET pin - return 0 directly in some simple routine to avoid variable initial - Change comments for function which was used by system - Change "ON" "OFF" to function str_on_off(...) - Change format of GUID - Add temp buffer for first efi.get_variable() - Free data if it was allocated - Change format of debug message of calibration date time print - Remove total_sz = 0, as it's not needed - Move save_calibration to after tuning_switch to avoid calibration data overwrite - Remove dev from structure tasdevice_hda, use dev in tasdevice_priv - Remove 0 from { "tas2781-hda", 0}, & {"TXNW2781", 0}, - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v3: - Move HID up to above /* Non-conforming _HID ... */ in scan.c, for avoid misunderstanding. - Move HID up to above /* Non-conforming _HID ... */ in serial-multi-instantiate.c, for avoid misunderstanding. - Change objs to y for snd-hda-scodec-tas2781-spi- in Makefile. - Change format for some define in tas2781-spi.h. - Add define for TASDEVICE_MAX_BOOK_NUM, TASDEVICE_MAX_PAGE. - Move declare of "enum calib_data {" to header file from source file. - Remove "enum device_catlog_id {" as only one platform was supported now. - Remove "struct calidata", as we will save pure calibration data only. - Remove "struct calidata", "enum device_catlog_id" in tasdevice_priv. - Add calibration_data in tasdevice_priv to save pure calibration data. - Remove declare of tasdevice_save_calibration() and tasdevice_apply_calibration(), as it will be used within same c file. - Add below header files included in tas2781_hda_spi.c: bits.h, mutex.h, property.h, time.h, types.h. - Move two micro define to header file tas2781-spi.h. - Change format of some micro define. - Change format of some structure. - Remove irq in tas2781_hda, as have it already in tasdevice_priv. - Remove some local functions declare as not necessarily. - Return error if regmap_write() failed. - Change fixed value 2 to sizeof(data). - Remove all of EXPORT_SYMBOL_GPL, as all of function was called within same module. - Remove empty line after last return in some functions. - Remove some variable initialyzation. - Remove variable sub, store acpi_subsystem_id directly. - Remove wrong comments for device index, it's must, can't be NULL. - Remove some local variables, use elements in structure directly. - Append commas in last element of an array. - Change calibration data process, didn't save all of calibration data in EFI, get in local, and save pure calibration data in tasdevice_priv. - Call tasdevice_save_calibration() from function pointer in tasdevice_priv; - Remove subid as one platform was supported only, needn't check. - Add initialyzation for local variable. - Move regmap initialyzation to before acpi read. - Call tasdevice_apply_calibration() from function pointer in tasdevice_priv; - Remove MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB), as all functions was performed within same module. - Update format and variables declare order. - Change format of multi conditions for if. - Remove casting which is not needed. - Change variables type to avoid casting. - Remove Unneeded parentheses. - Change if check to avoid goto. - Add {} for keep same style. - Remove some local variables, use elements in structure directly. v2: - Change depends on SPI to SPI_MASTER in sound/pci/hda/Kconfig - Add select SND_HDA_SCODEC_COMPONENT in sound/pci/hda/Kconfig - Change comp_generic_fixup() 5th parameter from "-%s:00" to "-%s:00-tas2781-hda.%d" in sound/pci/hda/patch_realtek.c - Change chain_id from ALC285_FIXUP_THINKPAD_HEADSET_JACK to ALC285_FIXUP_HP_GPIO_LED in sound/pci/hda/patch_realtek.c - Remove project name "Gemtree" - Update information in file header of tas2781-spi.h. - Remove useless micro define. - Change TASDEVICE_I2CChecksum to TASDEVICE_Checksum - Remove enum control_bus as current code support SPI only. - Remove device define as current code support TAS2781 only. - Remove spi_device **spi_devs in structure tasdevice_priv. - Remove cal_binaryname in structure tasdevice_priv. - Remove ndev in structure tasdevice_priv. - Change isacpi and isspi, replace by control_bus(I2C or SPI). - Remove void tasdevice_spi_dsp_remove(void *context). - Add acpi_device and irq in structure tas2781_hda in tas2781_hda_spi.c. - Remove parameter chn in all of registers access functions. - Add tascodec_spi_init(). - Add chip select gpio set for SPI device. - Change tasdevice_tuning_switch() to tasdevice_spi_tuning_switch(). - Change device offset from tas_priv->ndev to tas_priv->index. - Change tasdevice_dsp_remove() to tasdevice_spi_dsp_remove(). - tasdevice_prmg_load() to tasdevice_spi_prmg_load(). - Change tasdevice_config_info_remove() to tasdevice_spi_config_info_remove(). - Add one dummy byte shift when read from page 2~127, or not book 0. - Change tasdevice_spi_change_chn_book() to tasdevice_spi_switch_book() - Perform put_device(physdev) before error return. - Change calibration data id to 2781 from 2783 in EFI - Change calibration data structure name in comments. - Remove file name in tas2781_spi_fwlib.c. - Remove #include <linux/i2c.h> as it not needed in SPI mode. - Change TAB to speace in micro define in tas2781_spi_fwlib.c. - Change to up-case for DSP. - Change all of multi-line comments format, empty first line. - Change all of sizeof(struct xx) to sizeof(*xx). - Remove Explicit casting for variables compare to avoid possible fault. - Return directly without goto. - Change 1 << xx to BIT(xx). - Remove deviceNumber[] as current SPI device will support one device only. - Add memory free before return error. - Change "buf[offset]; offset += 1" to buf[offset++]. - Remove some debug information print. - Change firmware binary to single device mode. - Change all bexx_to_cpup() to get_unaligned_bexx(). - Remove ndev process as current SPI device support single device only. - Remove chn and chnend for single device support only. - Change all of registers read/write function, remove parameter chn. - Create new function for single write, burst write, delay, field write in tasdevice_process_block(). --- drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 158 ++ sound/pci/hda/tas2781_hda_spi.c | 1273 +++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2007 +++++++++++++++++ 8 files changed, 3480 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 59771412686b..b1a8bf1043da 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1769,6 +1769,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 3be016cfe601..b15c819023d0 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -368,6 +368,17 @@ static const struct smi_node cs35l57_hda = { .bus_type = SMI_AUTO_DETECT, }; +static const struct smi_node tas2781_hda = { + .instances = { + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + {} + }, + .bus_type = SMI_AUTO_DETECT, +}; + /* * Note new device-ids must also be added to ignore_serial_bus_ids in * drivers/acpi/scan.c: acpi_device_enumeration_by_parent(). @@ -380,6 +391,7 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "CSC3556", (unsigned long)&cs35l56_hda }, { "CSC3557", (unsigned long)&cs35l57_hda }, { "INT3515", (unsigned long)&int3515_data }, + { "TXNW2781", (unsigned long)&tas2781_hda }, /* Non-conforming _HID for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, { "CLSA0101", (unsigned long)&cs35l41_hda }, diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250..62cd9ba408e2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -206,6 +206,20 @@ config SND_HDA_SCODEC_TAS2781_I2C comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m +config SND_HDA_SCODEC_TAS2781_SPI + tristate "Build TAS2781 HD-audio side codec support for SPI Bus" + depends on SPI_MASTER + depends on ACPI + depends on EFI + depends on SND_SOC + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 SPI HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_SPI=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 80210f845df2..210c406dfbc5 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -40,6 +40,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -72,6 +73,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_SPI) += snd-hda-scodec-tas2781-spi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba0ce8750ca4..996e930b4cef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6926,6 +6926,11 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } +static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7504,6 +7509,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, @@ -9733,6 +9739,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC245_FIXUP_TAS2781_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_spi, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -10270,6 +10282,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8daa, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h new file mode 100644 index 000000000000..ecfc3c8bb821 --- /dev/null +++ b/sound/pci/hda/tas2781-spi.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for TAS2781 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2781_SPI_H__ +#define __TAS2781_SPI_H__ + +#define TASDEVICE_RATES \ + (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TASDEVICE_MAX_BOOK_NUM 256 +#define TASDEVICE_MAX_PAGE 256 + +#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) + +/* PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) +#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) +#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) +#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) +#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) +#define TASDEVICE_REG(book, page, reg) \ + (((book) << 16) | ((page) << 8) | ((reg) << 1)) + +/* Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET (0x19) +#define TAS2781_PRE_POST_RESET_CFG 3 + +/* Block Checksum */ +#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) + +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEVICE_CALIBRATION_DATA_SIZE 256 + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +struct tasdevice_priv { + struct tasdevice_fw *cali_data_fmw; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + bool force_fwload_status; + bool playback_started; + bool is_loading; + bool is_loaderr; + unsigned int cali_reg_array[CALIB_MAX]; + unsigned int cali_data[CALIB_MAX]; + unsigned int err_code; + void *codec; + int cur_book; + int cur_prog; + int cur_conf; + int fw_state; + int index; + int irq; + + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, + int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, + int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); +}; + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int *value); +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int value); +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, + unsigned int reg, unsigned int mask, + unsigned int value); + +void tasdevice_spi_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_spi_config_info_remove(void *context); +int tasdevice_spi_dsp_parser(void *context); +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_spi_dsp_remove(void *context); +void tasdevice_spi_calbin_remove(void *context); +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, + int rca_conf_no); +int tasdevice_spi_prmg_load(void *context, int prm_no); +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_spi_tuning_switch(void *context, int state); +int tas2781_spi_load_calibration(void *context, char *file_name, + unsigned short i); +#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c new file mode 100644 index 000000000000..a426b5bee451 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -0,0 +1,1272 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <linux/acpi.h> +#include <linux/array_size.h> +#include <linux/bits.h> +#include <linux/cleanup.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/pm_runtime.h> +#include <linux/property.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/time.h> +#include <linux/types.h> +#include <linux/units.h> + +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2781-spi.h" + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +struct tas2781_hda { + struct tasdevice_priv *priv; + struct acpi_device *dacpi; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *prof_ctl; +}; + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = TASDEVICE_MAX_SIZE, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = GENMASK(7, 0), + .selector_shift = 0, + .window_start = 0, + .window_len = TASDEVICE_MAX_PAGE, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .zero_flag_mask = true, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = TASDEVICE_MAX_SIZE, +}; + +static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) { + dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); + return ret; + } + tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); + } + return ret; +} + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int *val) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char data[2]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, sizeof(data)); + *val = data[1]; + } else { + ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + if (len > TASDEVICE_MAX_PAGE) + len = TASDEVICE_MAX_PAGE; + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[TASDEVICE_MAX_PAGE+1]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, + len + 1); + memcpy(data, buf + 1, len); + } else { + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, + len); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int mask, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret, val; + + /* + * In our TAS2781 SPI mode, read/write was masked in last bit of + * address, it cause regmap_update_bits() not work as expected. + */ + ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) +{ + int ret; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + fsleep(800); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } + ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); +} + +static int tascodec_spi_init(struct tasdevice_priv *tas_priv, + void *codec, struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret; + + /* + * Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + guard(mutex)(&tas_priv->codec_lock); + + ret = scnprintf(tas_priv->rca_binaryname, + sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->index); + if (ret <= 0) { + dev_err(tas_priv->dev, "rca name err:0x%08x\n", ret); + return ret; + } + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + return ret; +} + +static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) +{ + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + tas_priv->cur_book = -1; + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + /* Store default registers address for calibration data. */ + tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); + tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); + tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); + tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); + tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + + mutex_init(&tas_priv->codec_lock); +} + +static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int val, ret; + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_update_bits(tas_priv, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret, val; + + /* Read the primary device */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + return ret; + } + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int val, ret; + + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "set digital vol err in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + return ret; + } + + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2781_read_acpi(struct tas2781_hda *tas_hda, + const char *hid, + int id) +{ + struct tasdevice_priv *p = tas_hda->priv; + struct acpi_device *adev; + struct device *physdev; + u32 values[HDA_MAX_COMPONENTS]; + const char *property; + size_t nval; + int ret, i; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, "Failed to find ACPI device: %s\n", hid); + return -ENODEV; + } + + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + tas_hda->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + property = "ti,dev-index"; + ret = device_property_count_u32(physdev, property); + if (ret <= 0 || ret > ARRAY_SIZE(values)) { + ret = -EINVAL; + goto err; + } + nval = ret; + + ret = device_property_read_u32_array(physdev, property, values, nval); + if (ret) + goto err; + + p->index = U8_MAX; + for (i = 0; i < nval; i++) { + if (values[i] == id) { + p->index = i; + break; + } + } + if (p->index == U8_MAX) { + dev_dbg(p->dev, "No index found in %s\n", property); + ret = -ENODEV; + goto err; + } + + if (p->index == 0) { + /* All of amps share same RESET pin. */ + p->reset = devm_gpiod_get_index_optional(physdev, "reset", + p->index, GPIOD_OUT_LOW); + if (IS_ERR(p->reset)) { + dev_err_probe(p->dev, ret, "Failed on reset GPIO\n"); + goto err; + } + } + put_device(physdev); + + return 0; +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + acpi_dev_put(adev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + if (action == HDA_GEN_PCM_ACT_OPEN) { + pm_runtime_get_sync(dev); + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } else if (action == HDA_GEN_PCM_ACT_CLOSE) { + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->rcabin.ncfgs - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + if (tas_priv->rcabin.profile_cfg_id != val) { + tas_priv->rcabin.profile_cfg_id = val; + return 1; + } + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_priv->fmw->nr_programs - 1; + int val; + + val = clamp(nr_program, 0, max); + + if (tas_priv->cur_prog != val) { + tas_priv->cur_prog = val; + return 1; + } + + return 0; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->fmw->nr_configurations - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + + if (tas_priv->cur_conf != val) { + tas_priv->cur_conf = val; + return 1; + } + + return 0; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + * + * Return 0 if succeeded. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) { + change = false; + } else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 0", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 0", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 0", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 1", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 1", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 1", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl[] = { +{ + .name = "Speaker Profile Id - 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +{ + .name = "Speaker Profile Id - 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +}; +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl[] = { +{ + .name = "Speaker Program Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +{ + .name = "Speaker Program Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { +{ + .name = "Speaker Config Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +{ + .name = "Speaker Config Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + int i, rc; + + /* + * If no calibration data exist in tasdevice_priv *tas_priv, + * calibration apply will be ignored, and use default values + * in firmware binary, which was loaded during firmware download. + */ + if (tas_priv->cali_data[0] == 0) + return; + /* + * Calibration data was saved in tasdevice_priv *tas_priv as: + * unsigned int cali_data[CALIB_MAX]; + * and every data (in 4 bytes) will be saved in register which in + * book 0, and page number in page_array[], offset was saved in + * rgno_array[]. + */ + for (i = 0; i < CALIB_MAX; i++) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + tas_priv->cali_reg_array[i], + (unsigned char *)&tas_priv->cali_data[i], 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + tas_priv->index, i, rc); + } +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; + unsigned int attr, crc, offset, *tmp_val; + struct tm *tm = &tas_priv->tm; + unsigned long total_sz = 0; + efi_status_t status; + + tas_priv->cali_data[0] = 0; + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); + if (status == EFI_BUFFER_TOO_SMALL) { + if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &total_sz, data); + } + if (status != EFI_SUCCESS) + return -EINVAL; + + tmp_val = (unsigned int *)data; + if (tmp_val[0] == 2781) { + /* + * New features were added in calibrated Data V3: + * 1. Added calibration registers address define in + * a node, marked as Device id == 0x80. + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and + * uniqie_id for multi-link solutions + * 2. Support flexible number of devices instead of + * fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2781, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } // if Device number is 0x80, mean it's + * calibration registers address. + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ + crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; + + if (crc != tmp_val[3 + tmp_val[1] * 6]) + return 0; + + time64_to_tm(tmp_val[2], 0, tm); + for (int j = 0; j < tmp_val[1]; j++) { + offset = j * 6 + 3; + if (tmp_val[offset] == tas_priv->index) { + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[offset + i + 1]; + } else if (tmp_val[offset] == + TASDEVICE_CALIBRATION_REG_ADDRESS) { + for (int i = 0; i < CALIB_MAX; i++) { + buf = &data[(offset + i + 1) * 4]; + tas_priv->cali_reg_array[i] = + TASDEVICE_REG(buf[1], buf[2], + buf[3]); + } + } + tas_priv->apply_calibration(tas_priv); + } + } else { + /* + * Calibration data is in V1 format. + * struct cali_data { + * char cali_data[20]; + * } + * + * struct { + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) + * } ueft; + */ + crc = crc32(~0, data, 84) ^ ~0; + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[tas_priv->index * 5 + i]; + tas_priv->apply_calibration(tas_priv); + } + } + + return 0; +} + +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct hda_codec *codec = tas_priv->codec; + int i, j, ret; + + pm_runtime_get_sync(tas_priv->dev); + guard(mutex)(&tas_priv->codec_lock); + + ret = tasdevice_spi_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + /* Add control one time only. */ + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl[tas_priv->index].name, ret); + goto out; + } + j = tas_priv->index * ARRAY_SIZE(tas2781_snd_controls) / 2; + for (i = 0; i < 3; i++) { + tas_hda->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i+j], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i+tas_priv->index*3].name, + ret); + goto out; + } + } + + tasdevice_spi_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", + codec->core.subsystem_id, tas_priv->index); + ret = tasdevice_spi_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + /* Add control one time only. */ + tas_hda->dsp_prog_ctl = + snd_ctl_new1(&tas2781_dsp_prog_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_hda->dsp_conf_ctl = + snd_ctl_new1(&tas2781_dsp_conf_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl[tas_priv->index].name, ret); + goto out; + } + + /* Perform AMP reset before firmware download. */ + tas_priv->rcabin.profile_cfg_id = TAS2781_PRE_POST_RESET_CFG; + tasdevice_spi_tuning_switch(tas_priv, 0); + tas2781_spi_reset(tas_priv); + tasdevice_spi_tuning_switch(tas_priv, 1); + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + ret = tasdevice_spi_prmg_load(tas_priv, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "FW download failed = %d\n", ret); + goto out; + } + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; + + /* + * If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas_priv->save_calibration(tas_priv); + +out: + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_hda->priv->dev); + pm_runtime_put_autosuspend(tas_hda->priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + struct hda_codec *codec; + int ret; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (!comp) + return -EINVAL; + + if (comp->dev) + return -EBUSY; + + codec = parent->codec; + + pm_runtime_get_sync(dev); + + comp->dev = dev; + + strscpy(comp->name, dev_name(dev), sizeof(comp->name)); + + ret = tascodec_spi_init(tas_hda->priv, codec, THIS_MODULE, + tasdev_fw_ready); + if (!ret) + comp->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (comp && (comp->dev == dev)) { + comp->dev = NULL; + memset(comp->name, 0, sizeof(comp->name)); + comp->playback_hook = NULL; + } + + tas2781_hda_remove_controls(tas_hda); + + tasdevice_spi_config_info_remove(tas_hda->priv); + tasdevice_spi_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + component_del(tas_hda->priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_get_sync(tas_hda->priv->dev); + pm_runtime_disable(tas_hda->priv->dev); + + pm_runtime_put_noidle(tas_hda->priv->dev); + + mutex_destroy(&tas_hda->priv->codec_lock); +} + +static int tas2781_hda_spi_probe(struct spi_device *spi) +{ + struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; + const char *device_name; + int ret = 0; + + tas_hda = devm_kzalloc(&spi->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) + return -ENOMEM; + + spi->max_speed_hz = TAS2781_SPI_MAX_FREQ; + + tas_priv = devm_kzalloc(&spi->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + goto err; + tas_priv->dev = &spi->dev; + tas_hda->priv = tas_priv; + tas_priv->regmap = devm_regmap_init_spi(spi, &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate regmap: %d\n", + ret); + goto err; + } + if (strstr(dev_name(&spi->dev), "TXNW2781")) { + device_name = "TXNW2781"; + tas_priv->save_calibration = tas2781_save_calibration; + tas_priv->apply_calibration = tas2781_apply_calib; + } else { + goto err; + } + + tas_priv->irq = spi->irq; + dev_set_drvdata(&spi->dev, tas_hda); + ret = tas2781_read_acpi(tas_hda, device_name, + spi_get_chipselect(spi, 0)); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + tasdevice_spi_init(tas_priv); + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component fail: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&spi->dev); + + return ret; +} + +static void tas2781_hda_spi_remove(struct spi_device *spi) +{ + tas2781_hda_remove(&spi->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + return 0; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret, val; + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + guard(mutex)(&tas_hda->priv->codec_lock); + + ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, + &val); + if (ret < 0) + return ret; + + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + tas_hda->priv->cur_prog = -1; + + ret = tasdevice_spi_prmg_load(tas_hda->priv, 0); + if (ret < 0) { + dev_err(tas_hda->priv->dev, + "FW download failed = %d\n", ret); + return ret; + } + tas_hda->priv->apply_calibration(tas_hda->priv); + + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } + + return ret; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct spi_device_id tas2781_hda_spi_id[] = { + { "tas2781-hda", }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TXNW2781", }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct spi_driver tas2781_hda_spi_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_spi_id, + .probe = tas2781_hda_spi_probe, + .remove = tas2781_hda_spi_remove, +}; +module_spi_driver(tas2781_hda_spi_driver); + +MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); +MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c new file mode 100644 index 000000000000..c98f9beb77a0 --- /dev/null +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -0,0 +1,2006 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <asm/unaligned.h> +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> + +#include "tas2781-spi.h" + +#define OFFSET_ERROR_BIT BIT(31) + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/* should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +struct blktyp_devidx_map { + unsigned char blktyp; + unsigned char dev_idx; +}; + +/* fixed m68k compiling issue: mapping table can save code field */ +static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0x81 }, + { PRE_DEVICE_A_1X, 0x81 }, + { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0x82 }, + { PRE_DEVICE_B_1X, 0x82 }, + { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0x83 }, + { PRE_DEVICE_C_1X, 0x83 }, + { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0x84 }, + { PRE_DEVICE_D_1X, 0x84 }, + { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, + { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, +}; + +static const struct blktyp_devidx_map ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, +}; + +static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES, 0x80 }, + { MAIN_DEVICE_A, 0x81 }, + { COEFF_DEVICE_A, 0xC1 }, + { PRE_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B, 0x82 }, + { COEFF_DEVICE_B, 0xC2 }, + { PRE_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C, 0x83 }, + { COEFF_DEVICE_C, 0xC3 }, + { PRE_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D, 0x84 }, + { COEFF_DEVICE_D, 0xC4 }, + { PRE_DEVICE_D, 0xC4 }, +}; + +/* + * Device support different configurations for different scene, + * like voice, music, calibration, was write in regbin file. + * Will be stored into tas_priv after regbin was loaded. + */ +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* + * In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * DSP firmware. + */ + cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + return NULL; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if ((config_offset + 64) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto config_err; + } + config_offset += 64; + } + + if ((config_offset + 4) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto config_err; + } + + /* + * convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + /* + * Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(*bk_da), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto config_err; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + goto block_err; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + i++; + goto block_err; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + + return cfg_info; +block_err: + for (int j = 0; j < i; j++) + kfree(bk_da[j]); + kfree(bk_da); +config_err: + kfree(cfg_info); + return NULL; +} + +/* Regbin file parser function. */ +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + rca = &tas_priv->rcabin; + fw_hdr = &rca->fw_hdr; + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + buf = (unsigned char *)fmw->data; + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset++]; + fw_hdr->dev_family = buf[offset++]; + fw_hdr->reserve = buf[offset++]; + fw_hdr->ndev = buf[offset++]; + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", + fw_hdr->img_sz, total_config_sz, (int)offset); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -ENOMEM; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return ret; + } + offset += (int)fw_hdr->config_size[i]; + } + + return ret; +} + +/* fixed m68k compiling issue: mapping table can save code field */ +static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block) +{ + struct blktyp_devidx_map *p = + (struct blktyp_devidx_map *)non_ppc3_mapping_table; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + int i, n = ARRAY_SIZE(non_ppc3_mapping_table); + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; + n = ARRAY_SIZE(ppc3_tas2781_mapping_table); + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + p = (struct blktyp_devidx_map *)ppc3_mapping_table; + n = ARRAY_SIZE(ppc3_mapping_table); + } + + for (i = 0; i < n; i++) { + if (block->type == p[i].blktyp) { + dev_idx = p[i].dev_idx; + break; + } + } + + return dev_idx; +} + +/* Block parser function. */ +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host. + */ + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset++]; + block->pchksum = data[offset++]; + block->is_ychksum_present = data[offset++]; + block->ychksum = data[offset++]; + block->blk_size = get_unaligned_be32(&data[offset]); + offset += 4; + block->nr_subblocks = get_unaligned_be32(&data[offset]); + offset += 4; + + /* + * Fixed m68k compiling issue: + * 1. mapping table can save code field. + * 2. storing the dev_idx as a member of block can reduce unnecessary + * time and system resource comsumption of dev_idx mapping every + * time the block data writing to the dsp. + */ + block->dev_idx = map_dev_idx(tas_fmw, block); + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += block->blk_size; + + return offset; +} + +/* Data of block parser function. */ +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + img_data->nr_blk = get_unaligned_be32(&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data_kernel( + tas_fmw, blk, fmw, offset); + if (offset < 0) { + kfree(img_data->dev_blks); + return -EINVAL; + } + } + + return offset; +} + +/* Data of DSP program parser function. */ +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* skip 72 unused byts */ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* Data of DSP configurations parser function. */ +static int fw_parse_configuration_data_kernel(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* skip extra 16 bytes */ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* DSP firmware file header parser function for early PPC3 firmware binary. */ +static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_config *config; + struct tasdevice_prog *program; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + return -EINVAL; + } + offset += 2; + fw_hdr->device = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 2; + + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || + tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + return -EINVAL; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(*tas_fmw->programs), GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + program->prog_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); + offset += 4; + + /* + * The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) { + kfree(tas_fmw->programs); + return -ENOMEM; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &tas_fmw->configs[i]; + config->cfg_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + + return offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform single byte write to device. + */ +static int tasdevice_single_byte_wr(void *context, int dev_idx, + unsigned char *data, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int i, subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, "process_block: Out of boundary\n"); + return 0; + } + + for (i = 0; i < len; i++) { + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + dev_err(tas_priv->dev, + "process_block: single write error\n"); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 4; + } + + return subblk_offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform multi-write to device. + */ +static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (len % 4) { + dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", + __func__, len); + subblk_offset |= OFFSET_ERROR_BIT; + } + + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &data[subblk_offset + 4], len); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += (len + 4); + + return subblk_offset; +} + +/* Just delay for ms.*/ +static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned int sleep_time, subblk_offset = 2; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, "%s: delay Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + sleep_time = get_unaligned_be16(&data[2]) * 1000; + fsleep(sleep_time); + } + subblk_offset += 2; + + return subblk_offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform bits update. + */ +static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int rc, subblk_offset = 2; + + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_update_bits(tas_priv, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: update_bits error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 6; + + return subblk_offset; +} + +/* Data block process function. */ +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int blktyp = dev_idx & 0xC0, subblk_offset; + unsigned char subblk_typ = data[1]; + + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: + subblk_offset = tasdevice_single_byte_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_BURST: + subblk_offset = tasdevice_burst_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_DELAY: + subblk_offset = tasdevice_delay(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_FIELD_W: + subblk_offset = tasdevice_field_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + default: + subblk_offset = 2; + break; + } + if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + } else + tas_priv->cur_conf = -1; + } + subblk_offset &= ~OFFSET_ERROR_BIT; + + return subblk_offset; +} + +/* + * Device support different configurations for different scene, + * this function was used for choose different config. + */ +void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = pContext; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + unsigned int j, k; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < blk_data[j]->n_subblks; k++) { + tas_priv->is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} + +/* Block process function. */ +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + const unsigned int blk_size = block->blk_size; + unsigned char *data = block->data; + unsigned int i, length; + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + block->dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + return rc; + } + length += rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + rc = -ENOMEM; + return rc; + } + } + + return 0; +} + +/* DSP firmware file header parser function. */ +static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, + struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + offset += len; + + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + return -EINVAL; + } + offset += 4; + + fw_hdr->device = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 4; + fw_hdr->ndev = 1; + + return offset; +} + +/* DSP firmware file header parser function for size variabled header. */ +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + + return offset; +} + +/* DSP firmware file block parser function. */ +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + return -EINVAL; + } + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + return -EINVAL; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = get_unaligned_be32(&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += n; + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i, n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + return -EINVAL; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + return -EINVAL; + } + offset += n; + img_data->nr_blk = get_unaligned_be16(&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(*img_data->dev_blks), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) + return -EINVAL; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /* Not error in calibration Data file, return directly */ + dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + return offset; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), + GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &tas_fmw->programs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, + offset); + if (offset < 0) + return offset; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /* Not error for calibration Data file, return directly */ + return offset; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) + return -ENOMEM; + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + offset += n; + offset = fw_parse_data(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* yram5 page check. */ +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +/* DSP firmware yram block check. */ +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) { + in = check_inpage_yram_rg(cd, reg, len); + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) + in = check_inpage_yram_bk1(cd, page, reg, len); + else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +/* yram4 page check. */ +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +/* yram page check. */ +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (!in) + in = check_inblock_yram(cd, book, page, reg, len); + + return in; +} + +/* Checksum for data block. */ +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0, i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (len == 4)) { + /* DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_spi_dev_bulk_read(tasdevice, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) >= + TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) <= + (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + /* DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +/* Checksum for single register. */ +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + /* DSP swap command, pass */ + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + return 0; + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + return 0; + ret = tasdevice_spi_dev_read(tasdevice, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + return ret; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->err_code |= ERROR_YRAM_CRCCHK; + return -EAGAIN; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + + return ret; +} + +/* Block type check. */ +static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || + (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || + (type == MAIN_DEVICE_D)) + p->cur_prog = -1; + else + p->cur_conf = -1; +} + +/* Checksum for data bytes. */ +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, book, page, reg, val); + + if (ret > 0) { + *crc_chksum += ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, tas_priv); + +end: + return ret; +} + +/* Multi-data byte write. */ +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + + return ret; +} + +/* Checksum for block. */ +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); + set_err_prg_cfg(block->type, tas_priv); + return ret; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, tas_priv); + } else { + tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; + } + + return ret; +} + +/* Firmware block load function. */ +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int sleep_time, len, nr_cmds; + unsigned char offset, book, page, val; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_CHECKSUM, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /* Single byte write */ + if (offset <= 0x7F) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + break; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /* sleep command */ + if (offset == 0x81) { + /* book -- data[0] page -- data[1] */ + sleep_time = ((book << 8) + page)*1000; + fsleep(sleep_time); + continue; + } + /* Multiple bytes write */ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /* skip current blk */ + break; + } + + return ret; +} + +/* Firmware block load function. */ +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int ret = 0; + + block->nr_retry = 6; + if (tas_priv->is_loading == false) + return 0; + ret = tasdev_load_blk(tas_priv, block); + if (ret < 0) + dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); + + return ret; +} + +/* + * Select firmware binary parser & load callback functions by ppc3 version + * and firmware binary version. + */ +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else if (ppcver == 0x00) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + } else { + dev_err(tas_priv->dev, + "Wrong PPCVer :0x%08x\n", ppcver); + rc = -EINVAL; + } + } else { + dev_err(tas_priv->dev, "Wrong DrvVer : 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +/* DSP firmware binary file header parser function. */ +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); + offset += 72; + +out: + return offset; +} + +/* DSP firmware binary file parser function. */ +static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + return -EINVAL; + } + + tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) + return -EINVAL; + + fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + case 0x503: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + return ret; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + + return ret; +} + +/* DSP firmware binary file parser function. */ +int tasdevice_spi_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + return ret; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + + return ret; +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &prog->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +/* DSP firmware config block data remove function. */ +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &cfg->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +/* DSP firmware config remove function. */ +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +/* DSP firmware remove function. */ +void tasdevice_spi_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = context; + + if (!tas_dev->fmw) + return; + + if (tas_dev->fmw->programs) + tasdev_dsp_prog_remove(tas_dev->fmw->programs, + tas_dev->fmw->nr_programs); + if (tas_dev->fmw->configs) + tasdev_dsp_cfg_remove(tas_dev->fmw->configs, + tas_dev->fmw->nr_configurations); + kfree(tas_dev->fmw); + tas_dev->fmw = NULL; +} + +/* DSP firmware calibration data remove function. */ +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &tas_fmw->calibrations[i]; + if (!calibration) + continue; + + if (!calibration->dev_data.dev_blks) + continue; + + for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { + block = &calibration->dev_data.dev_blks[blks]; + if (!block) + continue; + kfree(block->data); + } + kfree(calibration->dev_data.dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +/* Calibration data from firmware remove function. */ +void tasdevice_spi_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + + if (tas_priv->cali_data_fmw) { + tas2781_clear_calfirmware(tas_priv->cali_data_fmw); + tas_priv->cali_data_fmw = NULL; + } +} + +/* Configuration remove function. */ +void tasdevice_spi_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **ci = rca->cfg_info; + unsigned int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} + +/* DSP firmware program block data load function. */ +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware program load interface function. */ +int tasdevice_spi_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int ret = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return -EINVAL; + } + if (prm_no >= 0 && prm_no <= tas_fmw->nr_programs) { + tas_priv->cur_conf = 0; + tas_priv->is_loading = true; + program = &tas_fmw->programs[prm_no]; + ret = tasdevice_load_data(tas_priv, &program->dev_data); + if (ret < 0) { + dev_err(tas_priv->dev, "Program failed %d.\n", ret); + return ret; + } + tas_priv->cur_prog = prm_no; + + conf = &tas_fmw->configs[tas_priv->cur_conf]; + ret = tasdevice_load_data(tas_priv, &conf->dev_data); + if (ret < 0) + dev_err(tas_priv->dev, "Config failed %d.\n", ret); + } else { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return -EINVAL; + } + + return ret; +} + +/* RCABIN configuration switch interface function. */ +void tasdevice_spi_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = context; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + else + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); +} -- 2.40.1

3 2

[PATCH] ASoC: topology: Fix route memory corruption
by Amadeusz Sławiński 15 Aug '24

15 Aug '24

It was reported that recent fix for memory corruption during topology load, causes corruption in other cases. Instead of being overeager with checking topology, assume that it is properly formatted and just duplicate strings. Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Closes: https://lore.kernel.org/linux-sound/171812236450.201359.3019210915105428447… Suggested-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0225bc5fc425a..4b166294602fa 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1052,21 +1052,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, break; } - route->source = devm_kmemdup(tplg->dev, elem->source, - min(strlen(elem->source), maxlen), - GFP_KERNEL); - route->sink = devm_kmemdup(tplg->dev, elem->sink, - min(strlen(elem->sink), maxlen), - GFP_KERNEL); + route->source = devm_kstrdup(tplg->dev, elem->source, GFP_KERNEL); + route->sink = devm_kstrdup(tplg->dev, elem->sink, GFP_KERNEL); if (!route->source || !route->sink) { ret = -ENOMEM; break; } if (strnlen(elem->control, maxlen) != 0) { - route->control = devm_kmemdup(tplg->dev, elem->control, - min(strlen(elem->control), maxlen), - GFP_KERNEL); + route->control = devm_kstrdup(tplg->dev, elem->control, GFP_KERNEL); if (!route->control) { ret = -ENOMEM; break; -- 2.34.1

5 6

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 15 Aug '24

15 Aug '24

> > To adapt to chrome, we have adjusted the buttons to match the system. > > This needs a bit more of a commit log explaining what's going on. > > > +#if IS_ENABLED(CONFIG_SND_SOC_MT8188_MT6359) > > + const int hs_keys[] = { > > + SND_JACK_BTN_0, SND_JACK_BTN_2, SND_JACK_BTN_3 > > + }; > > +#else > > + const int hs_keys[] = { > > + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2 > > + }; > > +#endif > > This will remap the buttons no matter what system we're running on so > while it's a bit better it's still going to cause issues. We could use > a per board approach like this but it needs to be at runtime, based on > the machine compatible rather than a build time selection. > > The support for this device was added in e794a894427b1d ("ASoC: > mediatek: mt8188-mt6359: add es8326 support") which was added in v6.8 > so this would also be a potential issue in mainline, given that this is > likely only used by Chromebooks and didn't make it to a LTS yet we could > possibly give it a go but we might have to revert. Because this change is to adapt mt8188-mt6359.On mt8188-mt6359, SND_JACK_BTN_2 corresponds to KEY_VOLUMEUP and SND_JACK_BTN_3 corresponds to KEY_VOLUMEDOWN. So we modified the es8326 driver to adapt the mt8188-mt6359. The changes we are making will not affect the application on all platforms.

1 0

[PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_stream_{lock/unlock}_irq()
by Kuninori Morimoto 15 Aug '24

15 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_{lock/unlock}_irq() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bad823718ae47..58381801e41d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,21 +49,9 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) -static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_unlock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) @@ -260,14 +248,14 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { dpcm_fe_dai_do_trigger(substream, fe->dpcm[stream].trigger_pending - 1); fe->dpcm[stream].trigger_pending = 0; } fe->dpcm[stream].runtime_update = state; - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); } static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, @@ -1272,10 +1260,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->be = be; dpcm->fe = fe; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(fe_substream); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(fe_substream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1320,11 +1308,12 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); LIST_HEAD(deleted_dpcms); snd_soc_dpcm_mutex_assert_held(fe); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", @@ -1343,7 +1332,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) list_del(&dpcm->list_be); list_move(&dpcm->list_fe, &deleted_dpcms); } - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); while (!list_empty(&deleted_dpcms)) { dpcm = list_first_entry(&deleted_dpcms, struct snd_soc_dpcm, -- 2.43.0

2 1

[PATCH 1/7] ASoC: fsl_audmix: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS()
by Fabio Estevam 15 Aug '24

15 Aug '24

From: Fabio Estevam <festevam(a)denx.de> Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl_audmix.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 1671a3037c60..f3a24758aedb 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -512,7 +512,6 @@ static void fsl_audmix_remove(struct platform_device *pdev) platform_device_unregister(priv->pdev); } -#ifdef CONFIG_PM static int fsl_audmix_runtime_resume(struct device *dev) { struct fsl_audmix *priv = dev_get_drvdata(dev); @@ -540,14 +539,11 @@ static int fsl_audmix_runtime_suspend(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_audmix_pm = { - SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, - fsl_audmix_runtime_resume, - NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, fsl_audmix_runtime_resume, + NULL) + SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static struct platform_driver fsl_audmix_driver = { @@ -556,7 +552,7 @@ static struct platform_driver fsl_audmix_driver = { .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, - .pm = &fsl_audmix_pm, + .pm = pm_ptr(&fsl_audmix_pm), }, }; module_platform_driver(fsl_audmix_driver); -- 2.34.1

2 7

Re: [PATCH] ASoC: sti: add missing probe entry for player and reader
by Mark Brown 14 Aug '24

14 Aug '24

On Mon, Jul 22, 2024 at 02:31:24PM +0200, jau(a)free.fr wrote: > Thanks for the review! > > See updated changlog, if ok for you, I will send a V2 Yes, that's fine.

1 0

[PATCH v1] ASoc: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 14 Aug '24

14 Aug '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header of tas2781-comlib.c - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c - Correct the filename in the header comments of tas2781-comlib.c, remove the wrong file name. - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - With all possible restrictions this can be on a single line besides the fact that the second one currently has a broken indentation. - remove unnecessary parentheses. - linux/* followed by asm/* as the latter is not so generic as the former. - leave the trailing comma - replace scnprintf with strscpy if necessary. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_restore for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - Add TASDEVICE_RCA_FW_OK to support only register setting bin in the device. - Add tas2563_dvc_table for relationship between tas2563 digtial gains and register values. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Add mutex into each kcontrol. - Rename tas2781_force_fwload_get, tas2781_force_fwload_put--> tasdev_force_fwload_get, tasdev_force_fwload_put - Define tas2781_snd_control and tas2563_snd_control for volume - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2563-tlv.h | 279 +++++++++ include/sound/tas2781-tlv.h | 260 --------- include/sound/tas2781.h | 72 +++ sound/soc/codecs/tas2781-comlib.c | 32 +- sound/soc/codecs/tas2781-fmwlib.c | 39 +- sound/soc/codecs/tas2781-i2c.c | 907 ++++++++++++++++++++++++++++-- 6 files changed, 1267 insertions(+), 322 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..2dab81a4c359 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0X7F, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 18161d02a96f..81fb34c9da89 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -23,6 +23,7 @@ #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 + #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) @@ -49,12 +50,69 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) +#define TAS2563_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x0f, 0x48) +#define TAS2563_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x0f, 0x40) +#define TAS2563_PRM_POW_REG TASDEVICE_REG(0x00, 0x0d, 0x3c) +#define TAS2563_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x10, 0x14) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_R0_REG TASDEVICE_REG(0x00, 0x17, 0x74) +#define TAS2781_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x18, 0x0c) +#define TAS2781_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x18, 0x14) +#define TAS2781_PRM_POW_REG TASDEVICE_REG(0x00, 0x13, 0x70) +#define TAS2781_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x18, 0x7c) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +128,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_restore; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -89,6 +155,9 @@ struct tasdevice_irqinfo { struct calidata { unsigned char *data; unsigned long total_sz; + unsigned int *reg_array; + unsigned int reg_array_sz; + unsigned int cali_dat_sz; }; struct tasdevice_priv { @@ -124,6 +193,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -150,6 +220,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..e7599f9fc125 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } @@ -89,6 +89,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 63626b982d04..2b5eb67a52e8 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2152,20 +2152,38 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + unsigned char *data = &cali_data->data[1]; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz + 1); + int j, rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (cal) + } + if (!priv->is_user_space_calidata) return; - - load_calib_data(priv, &cal->dev_data); + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + k++; + for (j = 0; j < cali_data->reg_array_sz; j++) { + rc = tasdevice_dev_bulk_write(priv, i, cali_data->reg_array[j], + &(data[k + 4 * j]), 4); + if (rc < 0) + dev_err(priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2260,9 +2278,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..e36934c8eeb2 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,9 +30,141 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_IDLE, + .val_len = 1, + .val = { 0x00 }, + }, + { + .reg = TAS2563_PRM_ENFF_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val_len = 4, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val_len = 4, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val_len = 4, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -49,6 +181,22 @@ static const struct of_device_id tasdevice_of_match[] = { MODULE_DEVICE_TABLE(of, tasdevice_of_match); #endif +static const int tas2563_cali_data_reg[] = { + TAS2563_PRM_R0_REG, + TAS2563_PRM_R0_LOW_REG, + TAS2563_PRM_INVR0_REG, + TAS2563_PRM_POW_REG, + TAS2563_PRM_TLIMIT_REG, +}; + +static const int tas2781_cali_data_reg[] = { + TAS2781_PRM_R0_REG, + TAS2781_PRM_R0_LOW_REG, + TAS2781_PRM_INVR0_REG, + TAS2781_PRM_POW_REG, + TAS2781_PRM_TLIMIT_REG, +}; + /** * tas2781_digital_getvol - get the volum control * @kcontrol: control pointer @@ -67,8 +215,13 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, @@ -78,8 +231,13 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, @@ -89,8 +247,13 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); + + return rc; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -101,8 +264,13 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); + + return rc; } static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, @@ -141,6 +309,414 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = tas_priv->cali_data.data; + + mutex_lock(&tas_priv->codec_lock); + if (!tas_priv->is_user_space_calidata || + tas_priv->cali_data.total_sz != bytes_ext->max) + goto out; + + memcpy(dst, data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst, int data_len) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + if (data_len != 4) + return rc; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, + unsigned char *dat) +{ + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + } + + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int rc = 1; + int i; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2781 && + bytes_ext->max != 9 * tas_priv->ndev) { + rc = 0; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + int k = i * 9; + + if (dat[k] != i) { + dev_err(tas_priv->dev, + "%s:no cal-setting for dev %d\n", __func__, i); + continue; + } + sngl_calib_start(tas_priv, i, dat + k); + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + unsigned char *data = cali_data->data; + int index = 0, i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id == TAS2563) + tas2563_calib_stop_put(tas_priv); + else + tas2781_calib_stop_put(tas_priv); + + for (i = 0; i < tas_priv->ndev; i++) { + data[index] = i; + index++; + for (j = 0; j < cali_data->reg_array_sz; j++) { + tasdevice_dev_bulk_read(tas_priv, i, + cali_data->reg_array[j], &data[index], 4); + index += 4; + } + } + mutex_unlock(&tas_priv->codec_lock); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *data = tas_priv->cali_data.data; + int rc = 1; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cali_data.total_sz != bytes_ext->max) { + rc = 0; + goto out; + } + tas_priv->is_user_space_calidata = true; + data[0] = bytes_ext->max; + memcpy(&data[1], ucontrol->value.bytes.data, bytes_ext->max - 1); +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + mutex_lock(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + dst[1] = val; + rc = 0; + break; + } + } + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_PRM_R0_REG; + else + reg = TAS2563_PRM_R0_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +817,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 5, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 5, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 5, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 5, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 5, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +836,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 2, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +847,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +869,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,29 +950,37 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; - int nr_controls = 1; + char *name; + int nr_controls = 2; int mix_index = 0; int ret; - char *name; prof_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(prof_ctrls[0]), GFP_KERNEL); - if (!prof_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!prof_ctrls) + return -ENOMEM; /* Create a mixer item for selecting the active profile */ name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!name) { - ret = -ENOMEM; - goto out; - } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); + if (!name) + return -ENOMEM; + + strscpy(name, "Speaker Profile Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -363,7 +991,6 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) ret = snd_soc_add_component_controls(tas_priv->codec, prof_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -out: return ret; } @@ -423,12 +1050,48 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + mutex_lock(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -437,23 +1100,23 @@ static int tasdevice_dsp_create_ctrls( */ dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(dsp_ctrls[0]), GFP_KERNEL); - if (!dsp_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!dsp_ctrls) + return -ENOMEM; /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, + active_dev_num = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + conf_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + prog_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!prog_name || !conf_name) { - ret = -ENOMEM; - goto out; - } + chip_id = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!active_dev_num || !conf_name || !chip_id || !prog_name) + return -ENOMEM; - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); + strscpy(prog_name, "Speaker Program Id", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -461,8 +1124,7 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + strscpy(conf_name, "Speaker Config Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -470,13 +1132,154 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + strscpy(active_dev_num, "Activate Tasdevice Num", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + strscpy(chip_id, "Tasdevice Chip Id", SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); -out: return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct tasdevice *tasdev = tas_priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + cali_data->reg_array = (unsigned int *)tas2781_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2781_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + cali_data->reg_array = (unsigned int *)tas2563_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2563_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (tas_priv->chip_id == TAS2781) + num_controls = 2; + else + num_controls = 1; + + /* + * Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + cali_ctrls = devm_kcalloc(tas_priv->dev, num_controls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + cali_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!cali_name || !ext_cali_data) + return -ENOMEM; + + strscpy(cali_name, "Speaker Calibrated Data", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz = cali_data->reg_array_sz * 4; + ext_cali_data->max = tas_priv->ndev * + (cali_data->cali_dat_sz + 1) + 1; + tas_priv->cali_data.total_sz = ext_cali_data->max; + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(tas_priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + cali_start_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!cali_start_name || !ext_cali_start) + return -ENOMEM; + + strscpy(cali_start_name, "Calibration Start", + SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + ext_cali_data->max = tas_priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls < i ? num_controls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -523,6 +1326,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -558,18 +1367,14 @@ static void tasdevice_fw_ready(const struct firmware *fmw, release_firmware(fmw); } -static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, int stream) { - struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int state = 0; /* Codec Lock Hold */ mutex_lock(&tas_priv->codec_lock); - if (event == SND_SOC_DAPM_PRE_PMD) - state = 1; - tasdevice_tuning_switch(tas_priv, state); + tasdevice_tuning_switch(tas_priv, mute); /* Codec Lock Release*/ mutex_unlock(&tas_priv->codec_lock); @@ -578,18 +1383,16 @@ static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, - 0, 0, tasdevice_dapm_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -668,11 +1471,12 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { .startup = tasdevice_startup, .hw_params = tasdevice_hw_params, .set_sysclk = tasdevice_set_dai_sysclk, + .mute_stream = tasdevice_mute, }; static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", @@ -724,6 +1528,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_restore); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.34.1

2 1

Re: [RFC PATCH 1/6] ALSA: compress: add Sample Rate Converter codec support
by Shengjiu Wang 14 Aug '24

14 Aug '24

On Wed, Aug 14, 2024 at 5:40 PM Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> wrote: > > > > Yes, to go further, I think we can use SND_AUDIOCODEC_PCM, then > > the SRC type will be dropped. > > sounds good. > > > But my understanding of the control means the .set_metadata() API, right? > > As I said, the output rate, output format, and ratio modifier are applied to > > the instances of ASRC, which is the snd_compr_stream in driver. > > so only the .set_metadata() API can be used for these purposes. > > Humm, this is more controversial. > > The term 'metadata' really referred to known information present in > headers or additional ID3 tags and not in the compressed file itself. > The .set_metadata was assumed to be called ONCE before decoding. > > But here you have a need to update the ratio modifier on a regular basis > to compensate for the drift. This isn't what this specific callback was > designed for. We could change and allow this callback to be used > multiple times, but then this could create problems for existing > implementations which cannot deal with modified metadata on the fly. .set_metadata can be called multi times now, no need to change currently. > > And then there's the problem of defining a 'key' for the metadata. the > definition of the key is a u32, so there's plenty of space for different > implementations, but a collision is possible. We'd need an agreement on > how to allocate keys to different solutions without changing the header > file for every implementation. Can we define a private space for each case? For example the key larger than 0x80000000 is private, each driver can define it by themself? > > It sounds like we'd need a 'runtime params' callback - unless there's a > better trick to tie the control and compress layers? >

3 2

[PATCH v1] ALSA: hda: tas2563: mark const variables as __maybe_unused
by Shenghao Ding 14 Aug '24

14 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> An earlier patch changed the DECLARE_TLV_DB_SCALE declaration, but now there are additional static const variables that cause the same build warnings: In file included from sound/pci/hda/tas2781_hda_i2c.c:23: include/sound/tas2563-tlv.h:21:28: error: 'tas2563_dvc_table' defined but not used [-Werror=unused-const-variable=] 21 | static const unsigned char tas2563_dvc_table[][4] = { | ^~~~~~~~~~~~~~~~~ ^~~~~~~~~~~~~~~ Mark it as unused as well. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.43.0

2 1

[PATCH v1] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 14 Aug '24

14 Aug '24

To adapt to chrome, we have adjusted the buttons to match the system. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..cf22d3b745d5 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -752,6 +752,16 @@ static void es8326_jack_button_handler(struct work_struct *work) static int button_to_report, press_count; static int prev_button, cur_button; +#if IS_ENABLED(CONFIG_SND_SOC_MT8188_MT6359) + const int hs_keys[] = { + SND_JACK_BTN_0, SND_JACK_BTN_2, SND_JACK_BTN_3 + }; +#else + const int hs_keys[] = { + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2 + }; +#endif + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ return; @@ -760,16 +770,16 @@ static void es8326_jack_button_handler(struct work_struct *work) switch (iface) { case 0x93: /* pause button detected */ - cur_button = SND_JACK_BTN_0; + cur_button = hs_keys[0]; break; case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + cur_button = hs_keys[1]; break; case 0x27: /* button volume down */ - cur_button = SND_JACK_BTN_2; + cur_button = hs_keys[2]; break; case 0x1e: case 0xe2: @@ -785,7 +795,7 @@ static void es8326_jack_button_handler(struct work_struct *work) if (press_count > 3) { /* report a press every 120ms */ snd_soc_jack_report(es8326->jack, cur_button, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); press_count = 0; } button_to_report = cur_button; @@ -800,9 +810,9 @@ static void es8326_jack_button_handler(struct work_struct *work) /* released or no pressed */ if (button_to_report != 0) { snd_soc_jack_report(es8326->jack, button_to_report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); button_to_report = 0; } es8326_disable_micbias(es8326->component); @@ -845,7 +855,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); -- 2.17.1

2 1

Behringer UCM204HD shows only both lines (4 speakers)
by GitHub issues - edited 14 Aug '24

14 Aug '24

alsa-project/alsa-ucm-conf issue #436 was edited from Udrys: Hi, I'm noob on linux so please don't beat me very hard :) I have Linux Mint 22 (Kernel 6.8.0-40-generic and Mint 21.3 had the same problem). So my UCM204HD shows only one 4 speaker channel. Not separate like on Windows or Debian. I tried to update configuration from this page but there are no difference. Maybe I'm missing something? Could you please help me? Thank you very much. P.s. Apparently it's absolutely doesn't matter what I do with /usr/share/alsa files, they don't change anything. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/436 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[tiwai-sound:for-linus 2/7] ld.lld: error: undefined symbol: i2c_acpi_client_count
by kernel test robot 14 Aug '24

14 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 8475a1d9bb7acf1cb15842dd24baab0e8ea4e4ff commit: 03c5c350e38d9346b69357d0e52c3c40495c14a0 [2/7] ALSA: hda/realtek: Add support for new HP G12 laptops config: i386-randconfig-013-20240811 (https://download.01.org/0day-ci/archive/20240814/202408141333.gYnaitcV-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240814/202408141333.gYnaitcV-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408141333.gYnaitcV-lkp@intel.com/ All errors (new ones prefixed by >>): >> ld.lld: error: undefined symbol: i2c_acpi_client_count >>> referenced by patch_realtek.c:6891 (sound/pci/hda/patch_realtek.c:6891) >>> sound/pci/hda/patch_realtek.o:(cs35lxx_autodet_fixup) in archive vmlinux.a -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 14 Aug '24

14 Aug '24

> > To adapt to chrome, we have adjusted the buttons to match the system. > > > case 0x4b: > > /* button volume up */ > > - cur_button = SND_JACK_BTN_1; > > + cur_button = SND_JACK_BTN_2; > > break; > > case 0x27: > > /* button volume down */ > > - cur_button = SND_JACK_BTN_2; > > + cur_button = SND_JACK_BTN_3; > > break; > > This is really problematic, we're changing the keys reported here in a > way that's not ABI compatible. If this was a new driver then fine but > this is quite old, it's in LTS releases, so we might be disrupting > people's existing systems. I'm not sure we have a good solution for > this at this point, we're not even swapping the buttons here but rather > using a different set of buttons. Probably ChromeOS needs to work > around this somehow, or we need to add some mechanism which allows > userspace to configure the driver mapping at runtime. We'll use a different method to solve this issue.thank you

1 0

[RESEND PATCH] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 13 Aug '24

13 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 313c17f76ca4..8c9efd9183ff 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -672,7 +672,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

2 1

[PATCH] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT
by Jerome Brunet 13 Aug '24

13 Aug '24

With PREEMPT_RT enabled, spinlocks become preemptible. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the upper half of a threaded irq handler won't be threaded and this causes scheduling problems if spinlocks are used in the upper half, like with regmap when '.fast_io' is set. In this particular instance, it is actually better to do everything in the bottom half and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@sa… Suggested-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..d0d05fa44d66 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the bottom half of a threaded irq with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- 2.43.0

3 4

[PATCH] ASoC: amd: acp: add legacy driver support for ACP7.1 based platforms
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Add acp pci driver and machine driver changes for ACP7.1 based platforms for legacy stack. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 7 ++++--- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-pci.c | 4 ++++ sound/soc/amd/acp/acp70.c | 12 ++++++++++-- sound/soc/amd/acp/amd.h | 1 + 6 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 04bd605fdce3..3cc083fac837 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -270,6 +270,7 @@ static int acp_power_on(struct acp_chip_info *chip) acp_pgfsm_ctrl_reg = ACP63_PGFSM_CONTROL; break; case ACP70_DEV: + case ACP71_DEV: acp_pgfsm_stat_reg = ACP70_PGFSM_STATUS; acp_pgfsm_ctrl_reg = ACP70_PGFSM_CONTROL; break; @@ -336,10 +337,9 @@ int acp_deinit(struct acp_chip_info *chip) if (ret) return ret; - if (chip->acp_rev != ACP70_DEV) + if (chip->acp_rev < ACP70_DEV) writel(0, chip->base + ACP_CONTROL); - - if (chip->acp_rev >= ACP70_DEV) + else writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } @@ -461,6 +461,7 @@ void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) check_acp6x_config(chip); break; case ACP70_DEV: + case ACP71_DEV: pdm_addr = ACP70_PDM_ADDR; check_acp70_config(chip); break; diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index a36300a4ed8a..e9ff4815c12c 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1766,7 +1766,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) } else if (drv_data->platform == ACP63) { links[i].platforms = platform_acp63_component; links[i].num_platforms = ARRAY_SIZE(platform_acp63_component); - } else if (drv_data->platform == ACP70) { + } else if ((drv_data->platform == ACP70) || (drv_data->platform == ACP71)) { links[i].platforms = platform_acp70_component; links[i].num_platforms = ARRAY_SIZE(platform_acp70_component); } else { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index a48546d8d407..93d9e3886b7e 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -56,6 +56,7 @@ enum platform_end_point { REMBRANDT, ACP63, ACP70, + ACP71, }; struct acp_mach_ops { diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index b0304b813cad..f7450a5bd103 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -95,6 +95,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_acp70"; chip->acp_rev = ACP70_DEV; break; + case 0x71: + chip->name = "acp_asoc_acp70"; + chip->acp_rev = ACP71_DEV; + break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); ret = -EINVAL; diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index a2cbdcca4313..81636faa22cd 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -147,7 +147,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) return -ENODEV; } - if (chip->acp_rev != ACP70_DEV) { + switch (chip->acp_rev) { + case ACP70_DEV: + case ACP71_DEV: + break; + default: dev_err(&pdev->dev, "Un-supported ACP Revision %d\n", chip->acp_rev); return -ENODEV; } @@ -178,7 +182,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) adata->num_dai = ARRAY_SIZE(acp70_dai); adata->rsrc = &rsrc; adata->machines = snd_soc_acpi_amd_acp70_acp_machines; - adata->platform = ACP70; + if (chip->acp_rev == ACP70_DEV) + adata->platform = ACP70; + else + adata->platform = ACP71; + adata->flag = chip->flag; acp_machine_select(adata); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index c095a34a7229..4f0cd7dd04e5 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -22,6 +22,7 @@ #define ACP6X_DEV 6 #define ACP63_DEV 0x63 #define ACP70_DEV 0x70 +#define ACP71_DEV 0x71 #define DMIC_INSTANCE 0x00 #define I2S_SP_INSTANCE 0x01 -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: Convert comma to semicolon
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Replace a comma between expression statements by a semicolon. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index f7e4af850309..08f368b3bbc8 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -657,9 +657,9 @@ static int mc_probe(struct platform_device *pdev) ctx->private = amd_ctx; card = &ctx->card; card->dev = &pdev->dev; - card->name = "amd-soundwire", - card->owner = THIS_MODULE, - card->late_probe = asoc_sdw_card_late_probe, + card->name = "amd-soundwire"; + card->owner = THIS_MODULE; + card->late_probe = asoc_sdw_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.34.1

2 1

[PATCH] ASoC: cs43130: Constify struct reg_sequence and reg_sequences
by Christophe JAILLET 13 Aug '24

13 Aug '24

'struct reg_sequence' and 'struct reg_sequences' are not modified in this drivers. Constifying these structures moves some data to a read-only section, so increase overall security. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o After: ===== text data bss dec hex filename 55562 6729 64 62355 f393 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. --- sound/soc/codecs/cs43130.c | 40 +++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cb4ca80f36d2..f8e2fb69ada2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1805,7 +1805,7 @@ static struct attribute *hpload_attrs[] = { }; ATTRIBUTE_GROUPS(hpload); -static struct reg_sequence hp_en_cal_seq[] = { +static const struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1820,7 +1820,7 @@ static struct reg_sequence hp_en_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_en_cal_seq2[] = { +static const struct reg_sequence hp_en_cal_seq2[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1828,7 +1828,7 @@ static struct reg_sequence hp_en_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_dis_cal_seq[] = { +static const struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_DXD1, 0x99}, {CS43130_DXD12, 0}, @@ -1836,12 +1836,12 @@ static struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dis_cal_seq2[] = { +static const struct reg_sequence hp_dis_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dc_ch_l_seq[] = { +static const struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1851,12 +1851,12 @@ static struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_l_seq2[] = { +static const struct reg_sequence hp_dc_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_r_seq[] = { +static const struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1866,12 +1866,12 @@ static struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_dc_ch_r_seq2[] = { +static const struct reg_sequence hp_dc_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_ac_ch_l_seq[] = { +static const struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1881,12 +1881,12 @@ static struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_l_seq2[] = { +static const struct reg_sequence hp_ac_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_r_seq[] = { +static const struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1896,24 +1896,24 @@ static struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_ac_ch_r_seq2[] = { +static const struct reg_sequence hp_ac_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_cln_seq[] = { +static const struct reg_sequence hp_cln_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, }; struct reg_sequences { - struct reg_sequence *seq; - int size; - unsigned int msk; + const struct reg_sequence *seq; + int size; + unsigned int msk; }; -static struct reg_sequences hpload_seq1[] = { +static const struct reg_sequences hpload_seq1[] = { { .seq = hp_en_cal_seq, .size = ARRAY_SIZE(hp_en_cal_seq), @@ -1951,7 +1951,7 @@ static struct reg_sequences hpload_seq1[] = { }, }; -static struct reg_sequences hpload_seq2[] = { +static const struct reg_sequences hpload_seq2[] = { { .seq = hp_en_cal_seq2, .size = ARRAY_SIZE(hp_en_cal_seq2), @@ -2041,7 +2041,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, } static int cs43130_hpload_proc(struct cs43130_private *cs43130, - struct reg_sequence *seq, int seq_size, + const struct reg_sequence *seq, int seq_size, unsigned int rslt_msk, int ac_idx) { int ret; @@ -2122,7 +2122,7 @@ static void cs43130_imp_meas(struct work_struct *wk) int i, ret, ac_idx; struct cs43130_private *cs43130; struct snd_soc_component *component; - struct reg_sequences *hpload_seq; + const struct reg_sequences *hpload_seq; cs43130 = container_of(wk, struct cs43130_private, work); component = cs43130->component; -- 2.45.2

2 1

[PATCH V2 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

The existing code modifies IRAM and DRAM size after sha dma start for vangogh platform. The problem with this sequence is that it might cause sha dma failure when firmware code binary size is greater than the default IRAM size. To fix this issue, Move the iram-dram fence register sequence prior to sha dma start. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes since v1: - Rephrase commit description sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

4 4

[PATCH 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

As per design, ACP iram-dram fence register sequence should be initiated before triggering SHA dma. This ensures that IRAM size will programmed correctly before initiaing SHA dma. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

4 4

[PATCH] ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers
by Krzysztof Kozlowski 13 Aug '24

13 Aug '24

There was no active maintenance from Banajit Goswami - last email is from 2019 - so make obvious that Qualcomm sound drivers are maintained by only one person. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Not moving to credits because lore does not point to significant contributions. Just ~140 emails: https://lore.kernel.org/all/?q=f%3A%22Banajit+Goswami%22 --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c8e16a649a0e..43e7668aacb0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18419,7 +18419,6 @@ F: drivers/crypto/intel/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> -M: Banajit Goswami <bgoswami(a)quicinc.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linux-arm-msm(a)vger.kernel.org S: Supported -- 2.43.0

2 2

[PATCH] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 13 Aug '24

13 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..8f9574189e1f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -673,7 +673,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

3 2

[PATCH] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 13 Aug '24

13 Aug '24

To adapt to chrome, we have adjusted the buttons to match the system. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..5b4586d9e67a 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -765,11 +765,11 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + cur_button = SND_JACK_BTN_2; break; case 0x27: /* button volume down */ - cur_button = SND_JACK_BTN_2; + cur_button = SND_JACK_BTN_3; break; case 0x1e: case 0xe2: @@ -785,7 +785,7 @@ static void es8326_jack_button_handler(struct work_struct *work) if (press_count > 3) { /* report a press every 120ms */ snd_soc_jack_report(es8326->jack, cur_button, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); press_count = 0; } button_to_report = cur_button; @@ -800,9 +800,9 @@ static void es8326_jack_button_handler(struct work_struct *work) /* released or no pressed */ if (button_to_report != 0) { snd_soc_jack_report(es8326->jack, button_to_report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); button_to_report = 0; } es8326_disable_micbias(es8326->component); @@ -845,7 +845,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); -- 2.17.1

2 1

[PATCH v2] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 13 Aug '24

13 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Change data type from int to __be32. v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 8a7fe48043d2..5a8193285f9f 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,22 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; + int offset = 0; int i, j, rc; + __be32 data; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.43.0

3 2

[PATCH 1/8] ASoC: SOF: amd: Fix for incorrect acp error satus register offset
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Adding 'dsp_intr_base' to ACP error status register offset in irq handler points to wrong register offset. ACP error status register offset got changed from ACP 6.0 onwards. Add 'acp_error_stat' descriptor field and update the value based on the ACP variant. Fixes: 0e44572a28a4 ("ASoC: SOF: amd: Add helper callbacks for ACP's DMA configuration") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp-dsp-offset.h | 3 ++- sound/soc/sof/amd/acp.c | 5 +++-- sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + sound/soc/sof/amd/pci-rmb.c | 1 + sound/soc/sof/amd/pci-rn.c | 1 + 6 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 59afbe2e0f42..66968efda869 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -76,7 +76,8 @@ #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 -#define ACP_ERROR_STATUS 0x18C4 +#define ACP3X_ERROR_STATUS 0x18C4 +#define ACP6X_ERROR_STATUS 0x1A4C #define ACP3X_AXI2DAGB_SEM_0 0x1880 #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..d0b7d1c54248 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -92,6 +92,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, unsigned int idx, unsigned int dscr_count) { struct snd_sof_dev *sdev = adata->dev; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int val, status; int ret; @@ -102,7 +103,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, val & (1 << ch), ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); if (ret < 0) { - status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_ERROR_STATUS); + status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->acp_error_stat); val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_ERR_STS_0 + ch * sizeof(u32)); dev_err(sdev->dev, "ACP_DMA_ERR_STS :0x%x ACP_ERROR_STATUS :0x%x\n", val, status); @@ -404,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW0_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW1_I2S_ERROR_REASON, 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_ERROR_STATUS, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index ec9170b3f068..6ac853ff6093 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -203,6 +203,7 @@ struct sof_amd_acp_desc { u32 probe_reg_offset; u32 reg_start_addr; u32 reg_end_addr; + u32 acp_error_stat; u32 sdw_max_link_count; u64 sdw_acpi_dev_addr; }; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index 54d42f83ce9e..c3da70549995 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -35,6 +35,7 @@ static const struct sof_amd_acp_desc acp63_chip_info = { .ext_intr_cntl = ACP6X_EXTERNAL_INTR_CNTL, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .ext_intr_stat1 = ACP6X_EXT_INTR_STAT1, + .acp_error_stat = ACP6X_ERROR_STATUS, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 4bc30951f8b0..194b7ff37e9e 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP6X_ERROR_STATUS, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index e08875bdfa8b..bff2d979ea6a 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc renoir_chip_info = { .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, .dsp_intr_base = ACP3X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP3X_ERROR_STATUS, .sram_pte_offset = ACP3X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP3X_AXI2DAGB_SEM_0, .acp_clkmux_sel = ACP3X_CLKMUX_SEL, -- 2.34.1

4 16

[PATCH 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 13 Aug '24

13 Aug '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: sma1307: Add bindings for Iron Device SMA1307 amplifier .../bindings/sound/irondevice,sma1307.yaml | 66 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2630 +++++++++++++++++ sound/soc/codecs/sma1307.h | 456 +++ 5 files changed, 3162 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

4 8

[PATCH RESEND 0/8] AMD SOF stack fixes and improvements
by Vijendar Mukunda 12 Aug '24

12 Aug '24

This patch series consists of few fixes and improvments for AMD SOF stack. Link: https://github.com/thesofproject/linux/pull/5103 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (8): ASoC: SOF: amd: Fix for incorrect acp error satus register offset ASoC: SOF: amd: fix for acp error reason registers wrong offset ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: fix for acp init sequence ASoC: SOF: amd: update conditional check for cache register update ASoC: SOF: amd: modify psp send command conditional checks ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure ASoC: amd: acp: Convert comma to semicolon sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +-- sound/soc/sof/amd/acp-dsp-offset.h | 6 ++- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 59 ++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 10 +++-- sound/soc/sof/amd/pci-acp63.c | 3 +- sound/soc/sof/amd/pci-rmb.c | 3 +- sound/soc/sof/amd/pci-rn.c | 3 +- sound/soc/sof/amd/pci-vangogh.c | 1 - 9 files changed, 60 insertions(+), 33 deletions(-) -- 2.34.1

2 9

[PATCH v1] ASoc: tas2781: Add new Kontrol to set tas2563 digital gain
by Shenghao Ding 11 Aug '24

11 Aug '24

Requriment from customer to add new kcontrol to set tas2563 digital gain and set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in tas2781-tlv.h. - Changed the copyright year to 2024 in tas2781.h. - Set "Speaker Force Firmware Load" as the common kcontrol for for both tas27871 and tas2563, that is tasdevice_snd_controls. - Only tas2781_snd_controls only for gain setting. - Add tas2563_dvc_tlv kcontrol for tas2563 digital gain setting. - In codec probe, for tas2563, loading tas2563 gain control; for tas2781, loading tas2781 gain control. --- include/sound/tas2781-tlv.h | 262 ++++++++++++++++++++++++++++++++- include/sound/tas2781.h | 3 +- sound/soc/codecs/tas2781-i2c.c | 135 ++++++++++++++++- 3 files changed, 392 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 1dc59005d241..99c41bfc7827 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -17,5 +17,265 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index cd8ce522b78e..3275575f5bfb 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -50,6 +50,7 @@ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) /* Volume control */ +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..4960ab6577f7 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +#include <asm/unaligned.h> static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, @@ -103,7 +104,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_putvol(tas_priv, ucontrol, mc); } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +119,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +128,114 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + /* find out the member same as or closer to the current volume */ + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +243,13 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -578,6 +680,27 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); @@ -605,8 +728,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

4 3

[PATCH v2] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 11 Aug '24

11 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change to Shenghao Ding and Baojun Xu, drop entries for the audio converter and add entries for both haptics drivers and middle- power audio amplifier drivers. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..d1f7e096b270 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,37 +22569,30 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c F: include/sound/tas2*.h -F: include/sound/tlv320*.h +F: include/sound/tas5086.h F: include/sound/tpa6130a2-plat.h F: sound/pci/hda/tas2781_hda_i2c.c -F: sound/soc/codecs/pcm1681.c -F: sound/soc/codecs/pcm1789*.* -F: sound/soc/codecs/pcm179x*.* -F: sound/soc/codecs/pcm186x*.* -F: sound/soc/codecs/pcm3008.* -F: sound/soc/codecs/pcm3060*.* -F: sound/soc/codecs/pcm3168a*.* -F: sound/soc/codecs/pcm5102a.c -F: sound/soc/codecs/pcm512x*.* F: sound/soc/codecs/tas2*.* -F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tas5*.* F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS -- 2.34.1

1 0

[PATCH 0/3] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 10 Aug '24

10 Aug '24

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/testing/devic… Two extra notes: 1. I don't quite understand whether the sound driver should have SoC-specific compatible or device-specific compatible. Some earlier patches by another author for a QCM6490 board and a QCS6490 board use device-specific compatible - but from what I can tell this is not how it's generally done for other sound drivers? 2. Unfortunately DisplayPort enablement itself for Fairphone 5 is not upstream yet. This is blocked by DSI display bringup upstream which is blocked by DSC 1:1:1 not being supported upstream yet and just working with a hacky patch. Nevertheless, DisplayPort audio was validated working with no additional sound-related changes so once DisplayPort gets enabled, sound should also just work upstream. Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Luca Weiss (3): ASoC: dt-bindings: qcom,sm8250: Add generic QCM6490 sound card ASoC: qcom: sc8280xp: Add support for generic QCM6490 arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 37 ++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 1 + 3 files changed, 39 insertions(+) --- base-commit: 6cdb38a56eaf615abc60fbeec0e4ccbdf93468e3 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

4 9

[PATCH AUTOSEL 5.10 02/11] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 10 Aug '24

10 Aug '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 25bf73a7e7bfa..ad3694d36d969 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -234,6 +234,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

2 2

[PATCH] ALSA: pcm: Add xrun counter for snd_pcm_substream
by Norman Bintang 10 Aug '24

10 Aug '24

This patch adds an xrun counter to snd_pcm_substream as an alternative to using logs from XRUN_DEBUG_BASIC. The counter provides a way to track the number of xrun occurences, accessible through the /proc interface. The counter is enabled when CONFIG_SND_PCM_XRUN_DEBUG is set. Example output: $ cat /proc/asound/card0/pcm9p/sub0/status owner_pid : 1425 trigger_time: 235.248957291 tstamp : 0.000000000 delay : 1912 avail : 480 avail_max : 1920 ----- hw_ptr : 672000 appl_ptr : 673440 xrun_counter: 3 # (new row) Signed-off-by: Norman Bintang <normanbt(a)chromium.org> Reviewed-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Tested-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Reviewed-by: David Riley <davidriley(a)chromium.org> --- include/sound/pcm.h | 3 +++ sound/core/pcm.c | 6 ++++++ sound/core/pcm_lib.c | 3 +++ 3 files changed, 12 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ac8f3aef9205..384032b6c59c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -498,6 +498,9 @@ struct snd_pcm_substream { /* misc flags */ unsigned int hw_opened: 1; unsigned int managed_buffer_alloc:1; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + unsigned int xrun_counter; /* number of times xrun happens */ +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index dc37f3508dc7..290690fc2abc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -462,6 +462,9 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + snd_iprintf(buffer, "xrun_counter: %d\n", substream->xrun_counter); +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -970,6 +973,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter = 0; +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e7905749c4a..6eaa950504cf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,6 +184,9 @@ void __snd_pcm_xrun(struct snd_pcm_substream *substream) pcm_warn(substream->pcm, "XRUN: %s\n", name); dump_stack_on_xrun(substream); } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter++; +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -- 2.46.0.76.ge559c4bf1a-goog

2 1

[PATCH 0/3] ALSA: update sample rate definition for eARC
by Jerome Brunet 10 Aug '24

10 Aug '24

This patchset updates some sample rate definition for eARC support. Notably it: * adds the definition for 128kHz * update the ASoC spdif codecs to include all supported rates, up to 768kHz * adds missing IEC958 definition for 128k, 352.4k, 384k and 705.6kHz On that last point, as noted in the related commit description, I'd like to stress that I have found the definitions using a testing equipement. I do not have access to the IEC958 specification. It would be best if someone with access to this specification could review/ack patch #2, before applying it. Jerome Brunet (3): ALSA: pcm: add support for 128kHz sample rate ALSA: IEC958 definition for consumer status channel update ASoC: spdif: extend supported rates to 768kHz include/sound/asoundef.h | 6 +++++- include/sound/pcm.h | 13 +++++++------ sound/core/pcm_native.c | 6 +++--- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- 5 files changed, 19 insertions(+), 12 deletions(-) -- 2.43.0

5 15

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - opened 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was opened from ViNi-Arco: What is the difference between `defaults.pcm.srate` and `defaults.pcm.rate`? I know there's `defaults.pcm.!rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - edited 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was edited from ViNi-Arco: What is the difference between `defaults.pcm.srate`? I know there's `defaults.pcm.!rate` and `defaults.pcm.rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoc: tas2781: Add Calibration Kcontrols and tas2563 digtial gain for Chromebook
by Shenghao Ding 09 Aug '24

09 Aug '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in the related files. - Add CAL_DAT_SZ for calibrated data size. - Add TAS2563_DVC_LVL for digtial gain kcontrol. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_restore for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - Add TASDEVICE_RCA_FW_OK to support only register setting bin in the device. - Add tas2563_dvc_table for relationship between tas2563 digtial gains and register values. - Correct the filename in the header comments of tas2781-comlib.c, tas2781-lib.c --> tas2781-comlib.c. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Add mutex into each kcontrol. - rename tas2781_force_fwload_get, tas2781_force_fwload_put--> tasdev_force_fwload_get, tasdev_force_fwload_put - rename tas2781_codec --> tasdev_codec - define tas2781_snd_control and tas2563_snd_control for volume - define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781-dsp.h | 1 + include/sound/tas2781.h | 92 ++- sound/soc/codecs/tas2781-comlib.c | 35 +- sound/soc/codecs/tas2781-fmwlib.c | 51 +- sound/soc/codecs/tas2781-i2c.c | 941 +++++++++++++++++++++++++++++- 5 files changed, 1061 insertions(+), 59 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..74ab6f113029 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -121,6 +121,7 @@ enum tasdevice_dsp_fw_state { TASDEVICE_DSP_FW_NONE = 0, TASDEVICE_DSP_FW_PENDING, TASDEVICE_DSP_FW_FAIL, + TASDEVICE_RCA_FW_OK, TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..2fb3d03d0040 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -23,6 +23,8 @@ #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 +#define CAL_DAT_SZ 20 + #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) @@ -31,6 +33,11 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + /*PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -43,21 +50,73 @@ (page * 128)) + reg) /*Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x00, 0X00, 0x01) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) /*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x00, 0x00, 0x7E) + +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) /* Volume control */ -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) -#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x00, 0x00, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x00, 0x00, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) -#define TASDEVICE_CMD_SING_W 0x1 -#define TASDEVICE_CMD_BURST 0x2 -#define TASDEVICE_CMD_DELAY 0x3 -#define TASDEVICE_CMD_FIELD_W 0x4 +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) +#define TAS2563_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x0f, 0x48) +#define TAS2563_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x0f, 0x40) +#define TAS2563_PRM_POW_REG TASDEVICE_REG(0x00, 0x0d, 0x3c) +#define TAS2563_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x10, 0x14) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_R0_REG TASDEVICE_REG(0x00, 0x17, 0x74) +#define TAS2781_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x18, 0x0c) +#define TAS2781_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x18, 0x14) +#define TAS2781_PRM_POW_REG TASDEVICE_REG(0x00, 0x13, 0x70) +#define TAS2781_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x18, 0x7c) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { TAS2563, @@ -69,7 +128,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_restore; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -88,6 +155,8 @@ struct tasdevice_irqinfo { struct calidata { unsigned char *data; unsigned long total_sz; + unsigned int *reg_array; + unsigned int reg_array_sz; }; struct tasdevice_priv { @@ -122,6 +191,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -148,6 +218,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 3aa81514dad7..9d7b56663b6b 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// tas2781-comlib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -89,6 +89,33 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdevice_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { @@ -254,8 +281,8 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) } else { for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, - TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); + TASDEVICE_REG_SWRESET, + TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, "dev %d swreset fail, %d\n", diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..19a5133083dc 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2153,20 +2153,39 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (CAL_DAT_SZ + 1); + int j, rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + if (!priv->is_user_space_calidata && + cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (cal) + } + if (!priv->is_user_space_calidata) return; - - load_calib_data(priv, &cal->dev_data); + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + for (j = 0; j < cali_data->reg_array_sz; j++) { + if (data[k] != j) + rc = tasdevice_dev_bulk_write(priv, i, + cali_data->reg_array[j], + &(data[k + 4 * j]), 4); + if (rc < 0) + dev_err(priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2261,9 +2280,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; @@ -2324,13 +2344,15 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* Only RCA file loaded still can work without speaker protection */ + if (!(tas_priv->fw_state == TASDEVICE_RCA_FW_OK || + tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK)) { + dev_err(tas_priv->dev, "No firmware loaded\n"); return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { /*dsp mode or tuning mode*/ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, @@ -2340,9 +2362,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..239f3538c558 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -13,6 +13,7 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // +#include <asm/unaligned.h> #include <linux/crc8.h> #include <linux/firmware.h> #include <linux/gpio/consumer.h> @@ -31,6 +32,122 @@ #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_PRM_ENFF_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -47,6 +164,22 @@ static const struct of_device_id tasdevice_of_match[] = { MODULE_DEVICE_TABLE(of, tasdevice_of_match); #endif +static const int tas2563_cali_data_reg[] = { + TAS2563_PRM_R0_REG, + TAS2563_PRM_R0_LOW_REG, + TAS2563_PRM_INVR0_REG, + TAS2563_PRM_POW_REG, + TAS2563_PRM_TLIMIT_REG, +}; + +static const int tas2781_cali_data_reg[] = { + TAS2781_PRM_R0_REG, + TAS2781_PRM_R0_LOW_REG, + TAS2781_PRM_INVR0_REG, + TAS2781_PRM_POW_REG, + TAS2781_PRM_TLIMIT_REG, +}; + /** * tas2781_digital_getvol - get the volum control * @kcontrol: control pointer @@ -65,8 +198,13 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, @@ -76,8 +214,13 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, @@ -87,8 +230,13 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -99,11 +247,16 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + return rc; } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +271,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +280,496 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = tas_priv->cali_data.data; + + mutex_lock(&tas_priv->codec_lock); + if (!tas_priv->is_user_space_calidata || + tas_priv->cali_data.total_sz != bytes_ext->max) { + goto out; + } + memcpy(dst, data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst, int data_len) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + if (data_len != 4) + return rc; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + rc = 0; + break; + } + } + + return rc; +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + unsigned char *dat = ucontrol->value.bytes.data; + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2781 && + bytes_ext->max != 8 * tas_priv->ndev) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + int k = i * 9; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + + if (dat[k] != i) { + dev_err(tas_priv->dev, + "%s: no cal-setting for dev %d\n", __func__, + i); + continue; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + &dat[i * 9 + 1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + &dat[i * 9 + 5], 4); + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *)tas2563_cali_start_reg[j].val, + 4); + } + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id == TAS2563) + tas2563_calib_stop_put(tas_priv); + else + tas2781_calib_stop_put(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *data = tas_priv->cali_data.data; + int rc = 1; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cali_data.total_sz != bytes_ext->max) { + rc = -1; + goto out; + } + tas_priv->is_user_space_calidata = true; + memcpy(data, ucontrol->value.bytes.data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + mutex_lock(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + dst[1] = val; + rc = 0; + break; + } + } + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_PRM_R0_REG; + else + reg = TAS2563_PRM_R0_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 5, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 5, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 5, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 5, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 5, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +777,22 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 2, tas2781_latch_reg_get, NULL), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), +}; + +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -167,6 +812,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -223,6 +893,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -316,15 +997,54 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_active_id_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_id_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int nr_dev = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + nr_dev = clamp(nr_dev, 0, max); + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_chn_switch(tas_priv, nr_dev); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { + char *active_dev_name, *cali_name, *chip_id, *conf_name, *prog_name; + struct calidata *cali_data = &tas_priv->cali_data; struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + struct soc_bytes_ext *ext_cali_data; + int nr_controls = 5; int mix_index = 0; int ret; + if (tas_priv->chip_id == TAS2781) + nr_controls++; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ @@ -336,11 +1056,20 @@ static int tasdevice_dsp_create_ctrls( } /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, + active_dev_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + cali_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + chip_id = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + prog_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!prog_name || !conf_name) { + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!active_dev_name || !cali_name || !conf_name || !chip_id || + !ext_cali_data || !prog_name) { ret = -ENOMEM; goto out; } @@ -363,6 +1092,68 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + scnprintf(active_dev_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Activate Tasdevice Id"); + dsp_ctrls[mix_index].name = active_dev_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_id; + dsp_ctrls[mix_index].get = tasdevice_active_id_get; + dsp_ctrls[mix_index].put = tasdevice_active_id_put; + mix_index++; + + scnprintf(chip_id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Tasdevice Chip Id"); + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + + scnprintf(cali_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Speaker Calibrated Data"); + ext_cali_data->max = tas_priv->ndev * CAL_DAT_SZ; + tas_priv->cali_data.total_sz = ext_cali_data->max; + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + dsp_ctrls[mix_index].name = cali_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].get = tasdev_cali_data_get; + dsp_ctrls[mix_index].put = tasdev_cali_data_put; + dsp_ctrls[mix_index].private_value = (unsigned long)ext_cali_data; + mix_index++; + + cali_data->data = devm_kzalloc(tas_priv->dev, tas_priv->ndev * + (cali_data->reg_array_sz * 4 + 1), GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + cali_start_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!cali_start_name || !ext_cali_start) { + ret = -ENOMEM; + goto out; + } + + scnprintf(cali_start_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Calibration Start"); + ext_cali_data->max = tas_priv->ndev * 9; + dsp_ctrls[mix_index].name = cali_start_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].put = tas2781_calib_start_put; + dsp_ctrls[mix_index].get = tasdev_nop_get; + dsp_ctrls[mix_index].private_value = + (unsigned long)ext_cali_start; + mix_index++; + } + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -370,6 +1161,54 @@ static int tasdevice_dsp_create_ctrls( return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + int rc; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali control err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + cali_data->reg_array = (unsigned int *)tas2781_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2781_cali_data_reg); + } else { + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + cali_data->reg_array = (unsigned int *)tas2563_cali_data_reg; + cali_data->reg_array_sz = sizeof(tas2563_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -380,23 +1219,36 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", tas_priv->dev_name); + ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } + + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +1269,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /*If DSP FW fail, DSP kcontrol won't be created */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +1317,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + if (!(tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK || + tas_priv->fw_state == TASDEVICE_RCA_FW_OK)) { + dev_err(tas_priv->dev, "Bin file not loaded\n"); + return -EINVAL; } - return ret; + return 0; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, @@ -541,7 +1392,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", @@ -565,13 +1416,41 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } + + rc = tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); - return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); + return rc; } static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_restore); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); @@ -591,8 +1470,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

5 6

[GIT PULL] ASoC fixes for v6.11-rc2
by Mark Brown 09 Aug '24

09 Aug '24

The following changes since commit 8400291e289ee6b2bf9779ff1c83a291501f017b: Linux 6.11-rc1 (2024-07-28 14:19:55 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc2 for you to fetch changes up to 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 Quite a lot of fixes have come in since the merge window, there's some repetitive fixes over the Qualcomm drivers increasing the patch count, along with a large batch of fixes from Cirrus. We also have some quirks and some individual fixes. ---------------------------------------------------------------- Bruno Ancona (1): ASoC: amd: yc: Support mic on HP 14-em0002la Charles Keepax (5): ASoC: cs42l43: Remove redundant semi-colon at end of function ASoC: soc-component: Add new snd_soc_component_get_kcontrol() helpers ASoC: cs35l45: Use new snd_soc_component_get_kcontrol_locked() helper ASoC: cs42l43: Cache shutter IRQ control pointers MAINTAINERS: Update Cirrus Logic parts to linux-sound mailing list Curtis Malainey (1): ASoC: SOF: Remove libraries from topology lookups Francesco Dolcini (1): ASoC: nau8822: Lower debug print priority Jerome Audu (1): ASoC: sti: add missing probe entry for player and reader Jerome Brunet (1): ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT Johan Hovold (1): ASoC: codecs: lpass-macro: fix missing codec version Krzysztof Kozlowski (10): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd934x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd938x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd939x: Correct reset GPIO polarity in example Krzysztof Stępniak (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E14 Gen 6 Mark Brown (3): ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports ASoC: fsl_micfil: Check the difference for i.MX8 and ASoC: Extend wm_adsp so cs35l56 can suppress controls Paul Handrigan (1): ASoC: cs530x: Change IN HPF Select kcontrol name Richard Fitzgerald (2): ASoC: cs-amp-lib: Fix NULL pointer crash if efi.get_variable is NULL ASoC: cs35l56: Handle OTP read latency over SoundWire Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms Simon Trimmer (3): ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value Takashi Iwai (1): ASoC: amd: yc: Add quirk entry for OMEN by HP Gaming Laptop 16-n0xxx Zhang Yi (1): ASoC: codecs: ES8326: button detect issue .../devicetree/bindings/sound/qcom,wcd934x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd939x.yaml | 4 +- MAINTAINERS | 4 +- include/sound/cs35l56.h | 5 ++ include/sound/soc-component.h | 5 ++ sound/soc/amd/yc/acp6x-mach.c | 21 ++++++ sound/soc/codecs/cs-amp-lib.c | 2 +- sound/soc/codecs/cs35l45.c | 11 +--- sound/soc/codecs/cs35l56-sdw.c | 77 ++++++++++++++++++++++ sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 11 ++++ sound/soc/codecs/cs42l43.c | 75 ++++++++++++++++----- sound/soc/codecs/cs42l43.h | 2 + sound/soc/codecs/cs530x.c | 8 +-- sound/soc/codecs/es8326.c | 2 + sound/soc/codecs/lpass-va-macro.c | 2 + sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/wcd937x-sdw.c | 4 +- sound/soc/codecs/wcd938x-sdw.c | 4 +- sound/soc/codecs/wcd939x-sdw.c | 4 +- sound/soc/codecs/wm_adsp.c | 17 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- sound/soc/fsl/fsl_micfil.c | 20 ++++-- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/meson/axg-fifo.c | 26 +++----- sound/soc/soc-component.c | 42 +++++++++--- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 36 files changed, 291 insertions(+), 84 deletions(-)

2 1

[PATCH 0/2] ASoC: codecs: warn on unknown codec version
by Dmitry Baryshkov 09 Aug '24

09 Aug '24

While debugging broken audio issues on some of Qualcomm platforms I stumbled upon the kernel not providing the actual error information. It prints an error from the wsa_macro driver, but the actual issue is in the VA macro driver. Add error message to point to the actual error location. va_macro 3370000.codec: Unknown VA Codec version, ID: 00 / 0f / 00 wsa_macro 3240000.codec: Unsupported Codec version (0) Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version sound/soc/codecs/lpass-macro-common.h | 6 ++++++ sound/soc/codecs/lpass-va-macro.c | 4 ++++ 2 files changed, 10 insertions(+) --- base-commit: 668d33c9ff922c4590c58754ab064aaf53c387dd change-id: 20240803-codec-version-db6f65ab51ce Best regards, -- Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org>

2 3

[tiwai-sound:for-next 69/83] sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/83] ALSA: vxpocket: Use standard print API config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090809.TAYiyrqJ-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c:207:8: error: no member named 'dev' in 'struct vx_core' 207 | chip->dev = &link->dev; | ~~~~ ^ >> sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ | p_dev include/linux/dev_printk.h:154:44: note: expanded from macro 'dev_err' 154 | dev_printk_index_wrap(_dev_err, KERN_ERR, dev, dev_fmt(fmt), ##__VA_ARGS__) | ^ include/linux/dev_printk.h:110:11: note: expanded from macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^ sound/pcmcia/vx/vxpocket.c:251:49: note: 'p_dev' declared here 251 | static int vxpocket_probe(struct pcmcia_device *p_dev) | ^ 2 errors generated. vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 73/83] sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared; did you mean 'chip'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: d41abde894830bfb77252653f606473728e930eb [73/83] ALSA: sparc: Use standard print API config: sparc64-defconfig (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) compiler: sparc64-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090648.J2EAijjH-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from include/linux/device.h:15, from include/linux/platform_device.h:13, from sound/sparc/cs4231.c:21: sound/sparc/cs4231.c: In function 'snd_cs4231_ebus_create': >> sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared (first use in this function); did you mean 'chip'? 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ sound/sparc/cs4231.c:2018:25: note: each undeclared identifier is reported only once for each function it appears in 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ vim +2018 sound/sparc/cs4231.c 1950 1951 static int snd_cs4231_ebus_create(struct snd_card *card, 1952 struct platform_device *op, 1953 int dev) 1954 { 1955 struct snd_cs4231 *chip = card->private_data; 1956 int err; 1957 1958 spin_lock_init(&chip->lock); 1959 spin_lock_init(&chip->c_dma.ebus_info.lock); 1960 spin_lock_init(&chip->p_dma.ebus_info.lock); 1961 mutex_init(&chip->mce_mutex); 1962 mutex_init(&chip->open_mutex); 1963 chip->flags |= CS4231_FLAG_EBUS; 1964 chip->op = op; 1965 memcpy(&chip->image, &snd_cs4231_original_image, 1966 sizeof(snd_cs4231_original_image)); 1967 strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); 1968 chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1969 chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; 1970 chip->c_dma.ebus_info.client_cookie = chip; 1971 chip->c_dma.ebus_info.irq = op->archdata.irqs[0]; 1972 strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); 1973 chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1974 chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; 1975 chip->p_dma.ebus_info.client_cookie = chip; 1976 chip->p_dma.ebus_info.irq = op->archdata.irqs[1]; 1977 1978 chip->p_dma.prepare = _ebus_dma_prepare; 1979 chip->p_dma.enable = _ebus_dma_enable; 1980 chip->p_dma.request = _ebus_dma_request; 1981 chip->p_dma.address = _ebus_dma_addr; 1982 1983 chip->c_dma.prepare = _ebus_dma_prepare; 1984 chip->c_dma.enable = _ebus_dma_enable; 1985 chip->c_dma.request = _ebus_dma_request; 1986 chip->c_dma.address = _ebus_dma_addr; 1987 1988 chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); 1989 chip->p_dma.ebus_info.regs = 1990 of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); 1991 chip->c_dma.ebus_info.regs = 1992 of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); 1993 if (!chip->port || !chip->p_dma.ebus_info.regs || 1994 !chip->c_dma.ebus_info.regs) { 1995 snd_cs4231_ebus_free(chip); 1996 dev_dbg(chip->card->dev, 1997 "cs4231-%d: Unable to map chip registers.\n", dev); 1998 return -EIO; 1999 } 2000 2001 if (ebus_dma_register(&chip->c_dma.ebus_info)) { 2002 snd_cs4231_ebus_free(chip); 2003 dev_dbg(chip->card->dev, 2004 "cs4231-%d: Unable to register EBUS capture DMA\n", 2005 dev); 2006 return -EBUSY; 2007 } 2008 if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) { 2009 snd_cs4231_ebus_free(chip); 2010 dev_dbg(chip->card->dev, 2011 "cs4231-%d: Unable to enable EBUS capture IRQ\n", 2012 dev); 2013 return -EBUSY; 2014 } 2015 2016 if (ebus_dma_register(&chip->p_dma.ebus_info)) { 2017 snd_cs4231_ebus_free(chip); > 2018 dev_dbg(chpi->card->dev, 2019 "cs4231-%d: Unable to register EBUS play DMA\n", 2020 dev); 2021 return -EBUSY; 2022 } 2023 if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) { 2024 snd_cs4231_ebus_free(chip); 2025 dev_dbg(chip->card->dev, 2026 "cs4231-%d: Unable to enable EBUS play IRQ\n", dev); 2027 return -EBUSY; 2028 } 2029 2030 if (snd_cs4231_probe(chip) < 0) { 2031 snd_cs4231_ebus_free(chip); 2032 return -ENODEV; 2033 } 2034 snd_cs4231_init(chip); 2035 2036 err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, 2037 chip, &snd_cs4231_ebus_dev_ops); 2038 if (err < 0) { 2039 snd_cs4231_ebus_free(chip); 2040 return err; 2041 } 2042 2043 return 0; 2044 } 2045 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 32/83] sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core'
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/83] ALSA: vx_core: Drop unused dev field config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090607.58bm8VK5-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core' 208 | chip->dev = &link->dev; | ~~~~ ^ 1 error generated. vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v2] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT
by Jerome Brunet 09 Aug '24

09 Aug '24

With PREEMPT_RT enabled a spinlock_t becomes a sleeping lock. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the primary handler won't be force-threaded and runs always in hardirq context. This is a problem because spinlock_t requires a preemptible context on PREEMPT_RT. In this particular instance, regmap mmio uses spinlock_t to protect the register access and IRQF_ONESHOT is set on the IRQ. In this case, it is actually better to do everything in threaded handler and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@sa… Suggested-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..75909196b769 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the threaded irq handler only with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- 2.43.0

3 2

[PATCH 1/4] ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example
by Krzysztof Kozlowski 09 Aug '24

09 Aug '24

The reset GPIO of WCD9370/WCD9375 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Cc: Alexey Klimov <alexey.klimov(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml index de397d879acc..f94203798f24 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml @@ -42,7 +42,7 @@ examples: pinctrl-names = "default", "sleep"; pinctrl-0 = <&wcd_reset_n>; pinctrl-1 = <&wcd_reset_n_sleep>; - reset-gpios = <&tlmm 83 GPIO_ACTIVE_HIGH>; + reset-gpios = <&tlmm 83 GPIO_ACTIVE_LOW>; vdd-buck-supply = <&vreg_l17b_1p8>; vdd-rxtx-supply = <&vreg_l18b_1p8>; vdd-px-supply = <&vreg_l18b_1p8>; -- 2.43.0

2 4

[PATCH v2 0/3] ASoC: tas*: Fix up GPIO usage
by Linus Walleij 09 Aug '24

09 Aug '24

The TI TAS drivers use some legacy GPIO code and headers, this series fixes it up. The TAS2781 is a special case since it adds a handful of lines of deviating code to reconfigure a GPIO line for IRQ mode and then never actually use the IRQ obtained in the code. Is the line used by autonomous hardware? I'm puzzled by this. Anyways the patch suggest how to solve this properly by fixing the parent irqchip and I'm happy to help. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased onto v6.11-rc1 - Link to v1: https://lore.kernel.org/r/20240701-asoc-tas-gpios-v1-0-d69ec5d79939@linaro.… --- Linus Walleij (3): ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 26 ++++---------------------- 9 files changed, 6 insertions(+), 37 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240701-asoc-tas-gpios-5c051d80d768 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 4

[PATCH] ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value
by Simon Trimmer 09 Aug '24

09 Aug '24

Device tuning files made with early revision tooling may contain configuration that can unmask IRQ signals that are owned by the host. Adding a safe default to the regmap patch ensures that the hardware matches the driver expectations. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..bd74fef33d49 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -36,6 +36,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, + { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, -- 2.43.0

2 1

[PATCH 0/2] ASoC: codecs: ES8326: button and input issue
by Zhang Yi 08 Aug '24

08 Aug '24

TO solve button and input issue, We have modified parts of the driver related to button and input. Zhang Yi (2): ASoC: codecs: ES8326: button detect issue ASoC: codecs: ES8326: input issue after init sound/soc/codecs/es8326.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.17.1

2 3

[PATCH] ASoC: amd: acp: add ZSC control register programming sequence
by Vijendar Mukunda 08 Aug '24

08 Aug '24

Add ZSC Control register programming sequence for ACP D0 and D3 state transitions for ACP7.0 onwards. This will allow ACP to enter low power state when ACP enters D3 state. When ACP enters D0 State, ZSC control should be disabled. Tested-by: Leo Li <sunpeng.li(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 5 +++++ sound/soc/amd/acp/amd.h | 2 ++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 4422cec81e3c..04bd605fdce3 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -321,6 +321,8 @@ int acp_init(struct acp_chip_info *chip) pr_err("ACP reset failed\n"); return ret; } + if (chip->acp_rev >= ACP70_DEV) + writel(0, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_init, SND_SOC_ACP_COMMON); @@ -336,6 +338,9 @@ int acp_deinit(struct acp_chip_info *chip) if (chip->acp_rev != ACP70_DEV) writel(0, chip->base + ACP_CONTROL); + + if (chip->acp_rev >= ACP70_DEV) + writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_deinit, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 87a4813783f9..c095a34a7229 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -103,6 +103,8 @@ #define ACP70_PGFSM_CONTROL ACP6X_PGFSM_CONTROL #define ACP70_PGFSM_STATUS ACP6X_PGFSM_STATUS +#define ACP_ZSC_DSP_CTRL 0x0001014 +#define ACP_ZSC_STS 0x0001018 #define ACP_SOFT_RST_DONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 0xffffffff -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver
by Vijendar Mukunda 08 Aug '24

08 Aug '24

As module device table added for AMD SoundWire machine driver MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD SoundWire machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 3419675e45a9..f7e4af850309 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -738,5 +738,4 @@ module_platform_driver(sof_sdw_driver); MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); MODULE_AUTHOR("Vijendar Mukunda <Vijendar.Mukunda(a)amd.com"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:amd_sof_sdw"); MODULE_IMPORT_NS(SND_SOC_SDW_UTILS); -- 2.34.1

4 3

[PATCH] ASoC: cs35l56: Use regmap_read_bypassed() to wake the device
by Simon Trimmer 08 Aug '24

08 Aug '24

Now that regmap_read_bypassed() has been added to the kernel it is preferable to wake the device with a read rather than a write as the utility function can be called at a time before the device has been identified. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..91b3c1c8575c 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -450,32 +450,23 @@ static const struct reg_sequence cs35l56_hibernate_seq[] = { REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_ALLOW_AUTO_HIBERNATE), }; -static const struct reg_sequence cs35l56_hibernate_wake_seq[] = { - REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_WAKEUP), -}; - static void cs35l56_issue_wake_event(struct cs35l56_base *cs35l56_base) { + unsigned int val; + /* * Dummy transactions to trigger I2C/SPI auto-wake. Issue two * transactions to meet the minimum required time from the rising edge * to the last falling edge of wake. * - * It uses bypassed write because we must wake the chip before + * It uses bypassed read because we must wake the chip before * disabling regmap cache-only. - * - * This can NAK on I2C which will terminate the write sequence so the - * single-write sequence is issued twice. */ - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); usleep_range(CS35L56_WAKE_HOLD_TIME_US, 2 * CS35L56_WAKE_HOLD_TIME_US); - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); cs35l56_wait_control_port_ready(); } -- 2.43.0

2 1

[tiwai-sound:for-next 66/82] sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: f8466d91f36d97e237ea1b3c6f19de980054a1b6 [66/82] ALSA: usx2y: Use standard print API config: x86_64-randconfig-161-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090249.8LE9qrae-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/usb/usx2y/us122l.c:15: >> sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used [-Wunused-but-set-variable] 668 | int frame = 0, iters = 0; | ^ 1 warning generated. vim +/iters +668 sound/usb/usx2y/usb_stream.c 030a07e441296c Karsten Wiese 2008-07-30 664 030a07e441296c Karsten Wiese 2008-07-30 665 int usb_stream_start(struct usb_stream_kernel *sk) 030a07e441296c Karsten Wiese 2008-07-30 666 { 030a07e441296c Karsten Wiese 2008-07-30 667 struct usb_stream *s = sk->s; 030a07e441296c Karsten Wiese 2008-07-30 @668 int frame = 0, iters = 0; 030a07e441296c Karsten Wiese 2008-07-30 669 int u, err; 030a07e441296c Karsten Wiese 2008-07-30 670 int try = 0; 030a07e441296c Karsten Wiese 2008-07-30 671 030a07e441296c Karsten Wiese 2008-07-30 672 if (s->state != usb_stream_stopped) 030a07e441296c Karsten Wiese 2008-07-30 673 return -EAGAIN; 030a07e441296c Karsten Wiese 2008-07-30 674 030a07e441296c Karsten Wiese 2008-07-30 675 subs_set_complete(sk->inurb, i_capture_start); 030a07e441296c Karsten Wiese 2008-07-30 676 subs_set_complete(sk->outurb, i_playback_start); 030a07e441296c Karsten Wiese 2008-07-30 677 memset(sk->write_page, 0, s->write_size); 030a07e441296c Karsten Wiese 2008-07-30 678 dotry: 030a07e441296c Karsten Wiese 2008-07-30 679 s->insize_done = 0; 030a07e441296c Karsten Wiese 2008-07-30 680 s->idle_insize = 0; 030a07e441296c Karsten Wiese 2008-07-30 681 s->idle_outsize = 0; 030a07e441296c Karsten Wiese 2008-07-30 682 s->sync_packet = -1; 030a07e441296c Karsten Wiese 2008-07-30 683 s->inpacket_head = -1; 030a07e441296c Karsten Wiese 2008-07-30 684 sk->iso_frame_balance = 0; 030a07e441296c Karsten Wiese 2008-07-30 685 ++try; 030a07e441296c Karsten Wiese 2008-07-30 686 for (u = 0; u < 2; u++) { 030a07e441296c Karsten Wiese 2008-07-30 687 struct urb *inurb = sk->inurb[u]; 030a07e441296c Karsten Wiese 2008-07-30 688 struct urb *outurb = sk->outurb[u]; 4c0a58ef36f3de Takashi Iwai 2021-05-17 689 030a07e441296c Karsten Wiese 2008-07-30 690 playback_prep_freqn(sk, outurb); 030a07e441296c Karsten Wiese 2008-07-30 691 inurb->number_of_packets = outurb->number_of_packets; 030a07e441296c Karsten Wiese 2008-07-30 692 inurb->transfer_buffer_length = 030a07e441296c Karsten Wiese 2008-07-30 693 inurb->number_of_packets * 030a07e441296c Karsten Wiese 2008-07-30 694 inurb->iso_frame_desc[0].length; d0f3a2eb906256 Karsten Wiese 2011-12-30 695 030a07e441296c Karsten Wiese 2008-07-30 696 if (u == 0) { 030a07e441296c Karsten Wiese 2008-07-30 697 int now; 030a07e441296c Karsten Wiese 2008-07-30 698 struct usb_device *dev = inurb->dev; 4c0a58ef36f3de Takashi Iwai 2021-05-17 699 030a07e441296c Karsten Wiese 2008-07-30 700 frame = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 701 do { 030a07e441296c Karsten Wiese 2008-07-30 702 now = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 703 ++iters; 030a07e441296c Karsten Wiese 2008-07-30 704 } while (now > -1 && now == frame); 030a07e441296c Karsten Wiese 2008-07-30 705 } 030a07e441296c Karsten Wiese 2008-07-30 706 err = usb_submit_urb(inurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 707 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 708 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 709 "%s: usb_submit_urb(sk->inurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 710 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 711 return err; 030a07e441296c Karsten Wiese 2008-07-30 712 } 030a07e441296c Karsten Wiese 2008-07-30 713 err = usb_submit_urb(outurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 714 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 715 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 716 "%s: usb_submit_urb(sk->outurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 717 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 718 return err; 030a07e441296c Karsten Wiese 2008-07-30 719 } d0f3a2eb906256 Karsten Wiese 2011-12-30 720 030a07e441296c Karsten Wiese 2008-07-30 721 if (inurb->start_frame != outurb->start_frame) { f8466d91f36d97 Takashi Iwai 2024-08-07 722 dev_dbg(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 723 "%s: u[%i] start_frames differ in:%u out:%u\n", f8466d91f36d97 Takashi Iwai 2024-08-07 724 __func__, u, inurb->start_frame, outurb->start_frame); 030a07e441296c Karsten Wiese 2008-07-30 725 goto check_retry; 030a07e441296c Karsten Wiese 2008-07-30 726 } 030a07e441296c Karsten Wiese 2008-07-30 727 } 030a07e441296c Karsten Wiese 2008-07-30 728 try = 0; 030a07e441296c Karsten Wiese 2008-07-30 729 check_retry: 030a07e441296c Karsten Wiese 2008-07-30 730 if (try) { 030a07e441296c Karsten Wiese 2008-07-30 731 usb_stream_stop(sk); 030a07e441296c Karsten Wiese 2008-07-30 732 if (try < 5) { 030a07e441296c Karsten Wiese 2008-07-30 733 msleep(1500); f8466d91f36d97 Takashi Iwai 2024-08-07 734 dev_dbg(&sk->dev->dev, "goto dotry;\n"); 030a07e441296c Karsten Wiese 2008-07-30 735 goto dotry; 030a07e441296c Karsten Wiese 2008-07-30 736 } f8466d91f36d97 Takashi Iwai 2024-08-07 737 dev_warn(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 738 "%s: couldn't start all urbs on the same start_frame.\n", f8466d91f36d97 Takashi Iwai 2024-08-07 739 __func__); 030a07e441296c Karsten Wiese 2008-07-30 740 return -EFAULT; 030a07e441296c Karsten Wiese 2008-07-30 741 } 030a07e441296c Karsten Wiese 2008-07-30 742 030a07e441296c Karsten Wiese 2008-07-30 743 sk->idle_inurb = sk->inurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 744 sk->idle_outurb = sk->outurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 745 sk->completed_inurb = sk->inurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 746 sk->completed_outurb = sk->outurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 747 :::::: The code at line 668 was first introduced by commit :::::: 030a07e441296c372f946cd4065b5d831d8dc40c ALSA: Add USB US122L driver :::::: TO: Karsten Wiese <fzu(a)wemgehoertderstaat.de> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 69/82] sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared; did you mean 'p_dev'?
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/82] ALSA: vxpocket: Use standard print API config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090110.t0mWbTyh-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': sound/pcmcia/vx/vxpocket.c:207:13: error: 'struct vx_core' has no member named 'dev' 207 | chip->dev = &link->dev; | ^~ In file included from include/linux/device.h:15, from include/sound/core.h:10, from sound/pcmcia/vx/vxpocket.c:13: sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_probe': >> sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared (first use in this function); did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ sound/pcmcia/vx/vxpocket.c:273:26: note: each undeclared identifier is reported only once for each function it appears in 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ALSA: snd-usb-caiaq: use snd_pcm_rate_to_rate_bit
by Jerome Brunet 08 Aug '24

08 Aug '24

Use snd_pcm_rate_to_rate_bit() helper provided by Alsa instead re-implementing it. This reduce code duplication and helps when changing some Alsa definition is necessary. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/usb/caiaq/audio.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4981753652a7..e62a4ea1d19c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -174,14 +174,6 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) return 0; } -/* this should probably go upstream */ -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 -#error "Change this table" -#endif - -static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; - static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { int bytes_per_sample, bpp, ret, i; @@ -233,10 +225,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ - for (i=0; i < ARRAY_SIZE(rates); i++) - if (runtime->rate == rates[i]) - cdev->pcm_info.rates = 1 << i; - + cdev->pcm_info.rates = snd_pcm_rate_to_rate_bit(runtime->rate); snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; -- 2.43.0

2 1

[tiwai-sound:for-next 32/79] sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev'
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b88d0890ed9adab413ea991fac90842688e1017 commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/79] ALSA: vx_core: Drop unused dev field config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090016.kW0TA6fc-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': >> sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev' 208 | chip->dev = &link->dev; | ^~ vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 08 Aug '24

08 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support" Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: SOF: Intel: add initial support for PTL ASoC: Intel: soc-acpi: add PTL match tables include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 121 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 +++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 ++- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 9

[PATCH 2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 07 Aug '24

07 Aug '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a51..757ade6373ed 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1095,6 +1095,11 @@ int cs35l56_system_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_resume); +static int cs35l56_control_add_nop(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl) +{ + return 0; +} + static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) { struct wm_adsp *dsp; @@ -1117,6 +1122,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp->fw = 12; dsp->wmfw_optional = true; + /* + * None of the firmware controls need to be exported so add a no-op + * callback that suppresses creating an ALSA control. + */ + dsp->control_add = &cs35l56_control_add_nop; + dev_dbg(cs35l56->base.dev, "DSP system name: '%s'\n", dsp->system_name); ret = wm_halo_init(dsp); -- 2.43.0

1 0

[PATCH alsa-lib] seq: Add snd_seq_{get|set}_ump_is_midi1() API functions
by Takashi Iwai 07 Aug '24

07 Aug '24

Implement the API function calls corresponding to the new sequencer port flag bit that has been added recently to the kernel. A UMP MIDI 2.0 device allow to have an optional MIDI 1.0 port while speaking in MIDI 2.0 protocol for other UMP Groups. The new seq port flag indicates that. This is rather a minor difference, and since ALSA sequencer core covers the all conversions, usually you don't have to worry about it at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 2 ++ include/sound/uapi/asequencer.h | 2 ++ src/Versions.in.in | 2 ++ src/seq/seq.c | 29 +++++++++++++++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/seq.h b/include/seq.h index 4f03df438288..2eee95a643b4 100644 --- a/include/seq.h +++ b/include/seq.h @@ -318,6 +318,7 @@ int snd_seq_port_info_get_timestamp_real(const snd_seq_port_info_t *info); int snd_seq_port_info_get_timestamp_queue(const snd_seq_port_info_t *info); int snd_seq_port_info_get_direction(const snd_seq_port_info_t *info); int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info); +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info); void snd_seq_port_info_set_client(snd_seq_port_info_t *info, int client); void snd_seq_port_info_set_port(snd_seq_port_info_t *info, int port); @@ -334,6 +335,7 @@ void snd_seq_port_info_set_timestamp_real(snd_seq_port_info_t *info, int realtim void snd_seq_port_info_set_timestamp_queue(snd_seq_port_info_t *info, int queue); void snd_seq_port_info_set_direction(snd_seq_port_info_t *info, int direction); void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group); +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1); int snd_seq_create_port(snd_seq_t *handle, snd_seq_port_info_t *info); int snd_seq_delete_port(snd_seq_t *handle, int port); diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index 923dfdddfc76..b3e9df5894af 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -477,6 +477,8 @@ struct snd_seq_remove_events { #define SNDRV_SEQ_PORT_FLG_TIMESTAMP (1<<1) #define SNDRV_SEQ_PORT_FLG_TIME_REAL (1<<2) +#define SNDRV_SEQ_PORT_FLG_IS_MIDI1 (1<<3) /* Keep MIDI 1.0 protocol */ + /* port direction */ #define SNDRV_SEQ_PORT_DIR_UNKNOWN 0 #define SNDRV_SEQ_PORT_DIR_INPUT 1 diff --git a/src/Versions.in.in b/src/Versions.in.in index 90849277c983..298e610c2099 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -209,6 +209,8 @@ ALSA_1.2.13 { @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; @SYMBOL_PREFIX@snd_seq_has_tempo_base; + @SYMBOL_PREFIX@snd_seq_port_info_get_ump_is_midi1; + @SYMBOL_PREFIX@snd_seq_port_info_set_ump_is_midi1; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index d0ecedf0838f..347ff45544fa 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -2431,6 +2431,19 @@ int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info) return info->ump_group; } +/** + * \brief Get the status of the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \return 1 if it's an optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_set_ump_is_midi1() + */ +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info) +{ + assert(info); + return !!(info->flags & SNDRV_SEQ_PORT_FLG_IS_MIDI1); +} + /** * \brief Set the client id of a port_info container * \param info port_info container @@ -2635,6 +2648,22 @@ void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group) info->ump_group = ump_group; } +/** + * \brief Set the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \param is_midi1 non-zero for MIDI 1.0 port in MIDI 2.0 EP + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_get_ump_is_midi1() + */ +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1) +{ + assert(info); + if (is_midi1) + info->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; + else + info->flags &= ~SNDRV_SEQ_PORT_FLG_IS_MIDI1; +} + /** * \brief create a sequencer port on the current client * \param seq sequencer handle -- 2.43.0

1 0

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 07 Aug '24

07 Aug '24

Hi, > Based on the inputs/suggestions from Tudor, i am planning to add a new > layer between the SPI-NOR and MTD layers to support stacked and parallel > configurations. This new layer will be part of the spi-nor and located in > mtd/spi-nor/ Will AMD submit to maintain this layer? What happens if the maintainer will leave AMD? TBH, personally, I don't like to maintain such a niche feature. I'd really like to see some use cases and performance reports for this, like actual boards (and no evaluation boards don't count). Why wouldn't someone just use an octal flash? And as already mentioned there is also mtdcat, which seems to duplicate some features? -michael

5 13

[PATCH v1] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 07 Aug '24

07 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 49bd7097d..4dc3350d5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,21 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; - int i, j, rc; + int i, j, rc, data; + int offset = 0; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add Framework Laptop 13 (Intel Core Ultra) to quirks
by Dustin L. Howett 07 Aug '24

07 Aug '24

The Framework Laptop 13 (Intel Core Ultra) has an ALC285 that ships in a similar configuration to the ALC295 in previous models. It requires the same quirk for headset detection. --- Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- 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 1645d21d422f..480e82df7a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10678,6 +10678,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. --- base-commit: de9c2c66ad8e787abec7c9d7eff4f8c3cdd28aed change-id: 20240606-alsa-hda-realtek-add-framework-laptop-13-intel-core-ultra-to-quirks-2af5081e2f2c Best regards, -- Dustin L. Howett <dustin(a)howett.net>

2 1

[PATCH v2] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest asoc/for-6.12 branch include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9ddc..1edcb8d6f6eea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2388,14 +2388,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2525,26 +2517,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", snd_pcm_direction_name(stream), fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2558,10 +2536,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2605,23 +2580,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 1

Re: [PATCH 0/2] ASoC: Extend wm_adsp so cs35l56 can suppress controls
by Mark Brown 06 Aug '24

06 Aug '24

On Mon, 05 Aug 2024 10:27:19 +0000, Simon Trimmer wrote: > This pair of patches extend wm_adsp to add a callback that can be used > to control whether ALSA controls are added and then tweak cs35l56 to use > it to suppress controls made from firmware coefficients. > > Simon Trimmer (2): > ASoC: wm_adsp: Add control_add callback and export > wm_adsp_control_add() > ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() commit: 45b4acab4cac79503663f0a4be9eb3752db04d4b [2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients commit: 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de 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

1 0

[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 3

[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 122 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

2 3

[PATCH] ALSA: usb-audio: Add input gain and master output mixer controls for RME Babyface Pro
by Stefan Stistrup 06 Aug '24

06 Aug '24

Add missing input gain and master output mixer controls for RME Babyface Pro This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup <sstistrup(a)gmail.com> --- sound/usb/mixer_quirks.c | 162 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 161 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d771..26298e06b838 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,113 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + } else { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + } + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2906,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2938,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2997,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) { + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + } else { + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + } + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, -- 2.45.2

2 2

[RFC PATCH 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Aug '24

06 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. Shengjiu Wang (6): ALSA: compress: add Sample Rate Converter codec support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 + include/uapi/sound/compress_params.h | 9 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 769 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 10 files changed, 1284 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

5 23

[PATCH v1] ASoC: meson: axg-fifo: set option to use raw spinlock
by Arseniy Krasnov 06 Aug '24

06 Aug '24

Raw spinlock is needed here, because with enabled PREEMPT_RT, spinlock_t become preemptible, but this regmap lock could be acquired in IRQ handler. Found by lockdep: [ ] ============================= [ ] [ BUG: Invalid wait context ] [ ] 6.9.9-sdkernel #1 Tainted: G O [ ] ----------------------------- [ ] aplay/413 is trying to lock: [ ] ffff000003930018 (axg_fifo:356:(&axg_fifo_regmap_cfg)->lock){....}-{3:3},c [ ] other info that might help us debug this: [ ] context-{2:2} [ ] no locks held by aplay/413. [ ] stack backtrace: [ ] CPU: 0 PID: 413 Comm: aplay Tainted: G O 6.9.9-kernel #1 [ ] Hardware name: SberDevices SberBoom Mini (DT) [ ] Call trace: [ ] dump_backtrace+0x98/0xf0 [ ] show_stack+0x18/0x24 [ ] dump_stack_lvl+0x90/0xd0 [ ] dump_stack+0x18/0x24 [ ] __lock_acquire+0x9dc/0x1f10 [ ] lock_acquire.part.0+0xe8/0x228 [ ] lock_acquire+0x68/0x84 [ ] _raw_spin_lock_irqsave+0x60/0x88 [ ] regmap_lock_spinlock+0x18/0x2c [ ] regmap_read+0x3c/0x78 [ ] axg_fifo_pcm_irq_block+0x4c/0xc8 [ ] __handle_irq_event_percpu+0xa4/0x2f8 [ ] handle_irq_event+0x4c/0xbc [ ] handle_fasteoi_irq+0xa4/0x23c [ ] generic_handle_domain_irq+0x2c/0x44 [ ] gic_handle_irq+0x40/0xc4 [ ] call_on_irq_stack+0x24/0x4c [ ] do_interrupt_handler+0x80/0x84 [ ] el0_interrupt+0x5c/0x124 [ ] __el0_irq_handler_common+0x18/0x24 [ ] el0t_32_irq_handler+0x10/0x1c [ ] el0t_32_irq+0x194/0x198 Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-fifo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index ecb3eb7a9723d..a22298f74b35a 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -328,6 +328,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .val_bits = 32, .reg_stride = 4, .max_register = FIFO_CTRL2, + .use_raw_spinlock = true, }; int axg_fifo_probe(struct platform_device *pdev) -- 2.30.1

4 12

[PATCH] ASoC: codecs: lpass-macro: fix missing codec version
by Johan Hovold 05 Aug '24

05 Aug '24

Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- Cc: regressions(a)lists.linux.dev #regzbot introduced: 378918d59181 sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) -- 2.44.2

3 2

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 05 Aug '24

05 Aug '24

Hi, > > You get twice more capacity based on that configuration. I can't answer the > > second question because not working with field. But both of that configurations > > are used by customers. Adding Neal if he wants to add something more to it. > > Just to add a comment as I work directly with our customers. The main reason > this support is important is for our older SoCs, zynq and zynqmp. > > Most of our customers are using QSPI flash as the first boot memory to get > from the boot ROM to u-boot. They then typically use other memories, such as > eMMC for the Linux kernel, OS and file system. Agreed and that's probably the most prominent use case for NOR flashes anyway. > The issue we have on the zynq and zynqmp SoCs is that the boot ROM (code that > cannot be changed) will not boot from an OSPI flash. It will only boot from a > QSPI flash. This is what is forcing many of our customers down the QSPI path. > Since many of these customers are interested in additional speed and memory > size, they then end up using a parallel or stacked configuration because they > cannot use an OSPI with zynq or zynqmp. Above you've said, the bootloader is stored on the NOR flash and the bulk storage is eMMC. So why do you need bigger NOR flashes (where even the biggest NOR flash isn't enough)? I also don't understand "the boot ROM will just boot from QSPI". First, you cannot connect an octal flash anyway, because you only have an QSPI controller, right? Secondly, normally the bootrom will (at least) boot the first stage using normal (single line) SPI commands. Is that not the case for zynq and zynqmp? > All of our newer SoCs can boot from OSPI. I agree with you that if someone > could choose OSPI for performance, they would, so I do not expect parallel or > stacked configurations with our newer SoCs. Ok, but then the argument with bigger flashes are void, because you are back to be bound to one OSPI flash. > I get why you see this configuration as a niche, but for us, it is a very > large niche because zynq and zynqmp are two of our most successful SoC > families. Fair enough. But please find a way to support it without butchering the whole core. -michael

1 0

Re: [PATCH v2] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 17:50:44 +0200, Stefan Binding wrote: > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls. > > These controls were never intended to be exposed to users but the > firmware doesn't mark them hidden, so make the driver ignore them. > > Any changes in the coefficients handled by these controls needs to > be matched to the individual system by SSID, which is already handled > using the tuning file, when firmware is loaded, so UCM should not be > setting these controls anyway. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Applied to for-linus branch now. Thanks. Takashi

1 0

Re: [PATCH v3] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 16:31:39 +0200, Simon Trimmer wrote: > > A number of laptops have gone to market with old firmware versions that > export controls that have since been hidden, but we can't just install a > newer firmware because the firmware for each product is customized and > qualified by the OEM. The issue is that alsactl save and restore has no > idea what controls are good to persist which can lead to > misconfiguration. > > There is no reason that the UCM or user should need to interact with any > of the ALSA controls for the firmware coefficients so they can be > removed entirely, this also simplifies the driver. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > Changes in v3: > - Remove ability to add firmware coefficients as ALSA controls > entirely > > Changes in v2: > - v1 was accidentally the backport version for older kernels instead of > the one for v6.11 Applied now to for-linus branch. Thanks. Takashi

1 0

ucm2: sof-soundwire: add rt712 VA device
by GitHub pull_request - opened 05 Aug '24

05 Aug '24

alsa-project/alsa-ucm-conf pull request #435 was opened from shumingfan: Add support headphone/headset mic/speaker/dmic for rt712 VA Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

a52 plugin Unknown field card error
by GitHub issues - edited 04 Aug '24

04 Aug '24

alsa-project/alsa-lib issue #405 was edited from DocMAX: ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } a52:CARD=Adapter USB SPDIF Adapter Plugin to convert multichannel stream to A52 (AC3) bitstream [root@desktop ~]# speaker-test -c6 -Da52 speaker-test 1.2.12 Playback device is a52 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise ALSA lib pcm_rate.c:1701:(_snd_pcm_rate_open) Unknown field card Playback open error: -22,Invalid argument ``` how to fix this error? running arch linux all updated Issue URL : https://github.com/alsa-project/alsa-lib/issues/405 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

a52 plugin Unknown field card error
by GitHub issues - opened 04 Aug '24

04 Aug '24

alsa-project/alsa-lib issue #405 was opened from DocMAX: ``` a52:CARD=Adapter USB SPDIF Adapter Plugin to convert multichannel stream to A52 (AC3) bitstream [root@desktop ~]# speaker-test -c6 -Da52 speaker-test 1.2.12 Playback device is a52 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise ALSA lib pcm_rate.c:1701:(_snd_pcm_rate_open) Unknown field card Playback open error: -22,Invalid argument ``` how to fix this error? running arch linux all updated Issue URL : https://github.com/alsa-project/alsa-lib/issues/405 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 04 Aug '24

04 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change to Shenghao Ding and Baojun Xu, drop entries for the audio converter and add entries for both haptics drivers and middle- power audio amplifier drivers. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 8766f3e5e87e..34639103e463 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22567,37 +22567,30 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS AUDIO (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c F: include/sound/tas2*.h -F: include/sound/tlv320*.h +F: include/sound/tas5086.h F: include/sound/tpa6130a2-plat.h F: sound/pci/hda/tas2781_hda_i2c.c -F: sound/soc/codecs/pcm1681.c -F: sound/soc/codecs/pcm1789*.* -F: sound/soc/codecs/pcm179x*.* -F: sound/soc/codecs/pcm186x*.* -F: sound/soc/codecs/pcm3008.* -F: sound/soc/codecs/pcm3060*.* -F: sound/soc/codecs/pcm3168a*.* -F: sound/soc/codecs/pcm5102a.c -F: sound/soc/codecs/pcm512x*.* F: sound/soc/codecs/tas2*.* -F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tas5*.* F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS -- 2.34.1

1 0

[PATCH 0/4] ASoC: SOF: reshuffle and optimize structures
by Pierre-Louis Bossart 03 Aug '24

03 Aug '24

pahole reports a number of sub-optimal structure definitions with holes and cache-line alignment problems, this patchset optimizes a couple of structure frequently used. No functionality change, only re-ordering of structure members. Pierre-Louis Bossart (4): ASoC: SOF: sof-priv.h: optimize snd_sof_platform_stream_params ASoC: SOF: sof-priv.h: optimize snd_sof_mailbox ASoC: SOF: sof-priv.h: optimize snd_sof_ipc_msg ASoC: SOF: sof-audio.h: optimize snd_sof_pcm_stream_pipeline_list sound/soc/sof/sof-audio.h | 4 ++-- sound/soc/sof/sof-priv.h | 16 +++++++++------- 2 files changed, 11 insertions(+), 9 deletions(-) -- 2.43.0

2 5

[PATCH v4 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 03 Aug '24

03 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. All the dependencies required in the initial versions are now available in ASoc for-next branch. v4: topology name simplification for rt722 v3: add configuration with rt722 on Link3 v2: changed order of patches to fix git bisect error reported by Intel kbuild bot https://lore.kernel.org/oe-kbuild-all/202406131144.L6gW0I47-lkp@intel.com/ Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 145 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 ++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 ++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 290 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 6

[PATCH] ASoC: cs43130: Constify snd_soc_component_driver struct
by Christophe JAILLET 03 Aug '24

03 Aug '24

In order to constify `snd_soc_component_driver` struct, duplicate `soc_component_dev_cs43130` into a `soc_component_dev_cs43130_digital` and `soc_component_dev_cs43130_analog`. These 2 new structures share the same .dapm_widgets and .dapm_routes arrays but differ for .num_dapm_widgets and .num_dapm_routes. In the digital case, the last entries are not taken into account. Doing so has several advantages: - `snd_soc_component_driver` can be declared as const to move their declarations to read-only sections. - code in the probe is simpler. There is no need to concatenate some arrays to handle the "analog" case - this saves some memory because all_hp_widgets and analog_hp_routes can be removed. Before : ====== text data bss dec hex filename 53965 8265 4512 66742 104b6 sound/soc/codecs/cs43130.o After : ===== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. See discussion at [1]. [1]: https://lore.kernel.org/all/ZqNawRmAqBRLIoQq@opensource.cirrus.com/ --- sound/soc/codecs/cs43130.c | 73 +++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index be4037890fdb..cb4ca80f36d2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1415,7 +1415,7 @@ static const char * const bypass_mux_text[] = { static SOC_ENUM_SINGLE_DECL(bypass_enum, SND_SOC_NOPM, 0, bypass_mux_text); static const struct snd_kcontrol_new bypass_ctrl = SOC_DAPM_ENUM("Switch", bypass_enum); -static const struct snd_soc_dapm_widget digital_hp_widgets[] = { +static const struct snd_soc_dapm_widget hp_widgets[] = { SND_SOC_DAPM_MUX("Bypass Switch", SND_SOC_NOPM, 0, 0, &bypass_ctrl), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -1447,19 +1447,16 @@ static const struct snd_soc_dapm_widget digital_hp_widgets[] = { CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD)), -}; -static const struct snd_soc_dapm_widget analog_hp_widgets[] = { +/* Some devices have some extra analog widgets */ +#define NUM_ANALOG_WIDGETS 1 + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), }; -static struct snd_soc_dapm_widget all_hp_widgets[ - ARRAY_SIZE(digital_hp_widgets) + - ARRAY_SIZE(analog_hp_widgets)]; - -static const struct snd_soc_dapm_route digital_hp_routes[] = { +static const struct snd_soc_dapm_route hp_routes[] = { {"ASPIN PCM", NULL, "ASP PCM Playback"}, {"ASPIN DoP", NULL, "ASP DoP Playback"}, {"XSPIN DoP", NULL, "XSP DoP Playback"}, @@ -1472,15 +1469,12 @@ static const struct snd_soc_dapm_route digital_hp_routes[] = { {"Bypass Switch", "Internal", "HiFi DAC"}, {"HPOUTA", NULL, "Bypass Switch"}, {"HPOUTB", NULL, "Bypass Switch"}, -}; -static const struct snd_soc_dapm_route analog_hp_routes[] = { +/* Some devices have some extra analog routes */ +#define NUM_ANALOG_ROUTES 1 {"Bypass Switch", "Alternative", "Analog Playback"}, }; -static struct snd_soc_dapm_route all_hp_routes[ - ARRAY_SIZE(digital_hp_routes) + - ARRAY_SIZE(analog_hp_routes)]; static const unsigned int cs43130_asp_src_rates[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 @@ -2398,7 +2392,23 @@ static int cs43130_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_cs43130 = { +static const struct snd_soc_component_driver soc_component_dev_cs43130_digital = { + .probe = cs43130_probe, + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .set_sysclk = cs43130_component_set_sysclk, + .set_pll = cs43130_set_pll, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + /* Don't take into account the ending analog widgets and routes */ + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets) - NUM_ANALOG_WIDGETS, + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes) - NUM_ANALOG_ROUTES, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs43130_analog = { .probe = cs43130_probe, .controls = cs43130_snd_controls, .num_controls = ARRAY_SIZE(cs43130_snd_controls), @@ -2407,6 +2417,10 @@ static struct snd_soc_component_driver soc_component_dev_cs43130 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets), + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes), }; static const struct regmap_config cs43130_regmap = { @@ -2479,6 +2493,7 @@ static int cs43130_handle_device_data(struct cs43130_private *cs43130) static int cs43130_i2c_probe(struct i2c_client *client) { + const struct snd_soc_component_driver *component_driver; struct cs43130_private *cs43130; int ret; unsigned int reg; @@ -2596,39 +2611,15 @@ static int cs43130_i2c_probe(struct i2c_client *client) switch (cs43130->dev_id) { case CS43130_CHIP_ID: case CS43131_CHIP_ID: - memcpy(all_hp_widgets, digital_hp_widgets, - sizeof(digital_hp_widgets)); - memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), - analog_hp_widgets, sizeof(analog_hp_widgets)); - memcpy(all_hp_routes, digital_hp_routes, - sizeof(digital_hp_routes)); - memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), - analog_hp_routes, sizeof(analog_hp_routes)); - - soc_component_dev_cs43130.dapm_widgets = - all_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(all_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - all_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(all_hp_routes); + component_driver = &soc_component_dev_cs43130_analog; break; case CS43198_CHIP_ID: case CS4399_CHIP_ID: - soc_component_dev_cs43130.dapm_widgets = - digital_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(digital_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - digital_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(digital_hp_routes); + component_driver = &soc_component_dev_cs43130_digital; break; } - ret = devm_snd_soc_register_component(cs43130->dev, - &soc_component_dev_cs43130, + ret = devm_snd_soc_register_component(cs43130->dev, component_driver, cs43130_dai, ARRAY_SIZE(cs43130_dai)); if (ret < 0) { dev_err(cs43130->dev, -- 2.45.2

2 1

Soundblaster AE-5 100% Muted on alsamixer every boot!!
by GitHub issues - opened 02 Aug '24

02 Aug '24

alsa-project/alsa-lib issue #404 was opened from DavidRLTG: I am on Linux Mint Wilma (22), 64-bit. Obviously, I have a Soundblaster AE-5 Sound card The issue I am encountering is, well, every boot I have to enter Alsamixer and unmute every channel on my Soundblaster AE-5. alsactl store does nothing, and yes I ran it as superuser. Also, I'm on pipewire! I think Wilma comes with it preinstalled. sudo alsactl store is also temporary, the config is lost at every reboot, and doing sudo alsactl restore does nothing. Any ideas for a fix? Issue URL : https://github.com/alsa-project/alsa-lib/issues/404 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RESEND PATCH v2] ASoc: tas2781: Fix a compiling warning reported by robot kernel test due to adding tas2563_dvc_table
by Shenghao Ding 02 Aug '24

02 Aug '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Fixes: 75ed63a5ab5d ("ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Modify the submission subject v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..faa3e194f73b --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0X7F, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 00fd4d449ff3..d87263e43fdb 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

2 1

[PATCH -next] ASoC: fsl: lpc3xxx: Make some symbols static
by Yue Haibing 02 Aug '24

02 Aug '24

These symbols are not used outside of the files, make them static to fix sparse warnings: sound/soc/fsl/lpc3xxx-i2s.c:261:30: warning: symbol 'lpc3xxx_i2s_dai_ops' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-i2s.c:271:27: warning: symbol 'lpc3xxx_i2s_dai_driver' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-pcm.c:55:39: warning: symbol 'lpc3xxx_soc_platform_driver' was not declared. Should it be static? Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 ++-- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 62ef624d6dd4..bdb320bb3719 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -258,7 +258,7 @@ static int lpc3xxx_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { +static const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .probe = lpc3xxx_i2s_dai_probe, .startup = lpc3xxx_i2s_startup, .shutdown = lpc3xxx_i2s_shutdown, @@ -268,7 +268,7 @@ const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .set_fmt = lpc3xxx_i2s_set_dai_fmt, }; -struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { +static struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { .playback = { .channels_min = 1, .channels_max = 2, diff --git a/sound/soc/fsl/lpc3xxx-pcm.c b/sound/soc/fsl/lpc3xxx-pcm.c index c0d499b9b8ba..e6abaf63895a 100644 --- a/sound/soc/fsl/lpc3xxx-pcm.c +++ b/sound/soc/fsl/lpc3xxx-pcm.c @@ -52,7 +52,7 @@ static const struct snd_dmaengine_pcm_config lpc3xxx_dmaengine_pcm_config = { .prealloc_buffer_size = 128 * 1024, }; -const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { +static const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { .name = "lpc32xx-pcm", }; -- 2.34.1

2 1

[PATCH 00/12] ALSA/ASoC: use snd_pcm_direction_name()
by Kuninori Morimoto 02 Aug '24

02 Aug '24

Hi Tatasi, Mark Many drivers are selecting strings "playback" / "capture" by own handling, but we have snd_pcm_direction_name() function for it. This patch use it. One note is that snd_pcm_direction_name() will select "Playback" and "Capture", instead of "playback" / "capture". Almost all drivers are using it as dev_dbg() or dev_err() so no problem. But some other drivers are using it as other purpose. It might be issue (?). For example ASoC debugfs dir name will be changed by this patch. Kuninori Morimoto (12): ALSA: pci: pcxhr: use snd_pcm_direction_name() ALSA: pci: rme9652: use snd_pcm_direction_name() ALSA: trace: use snd_pcm_direction_name() ALSA: aloop: use snd_pcm_direction_name() ALSA: pcm_timer: use snd_pcm_direction_name() ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() include/trace/events/asoc.h | 3 ++- sound/core/pcm_timer.c | 3 +-- sound/drivers/aloop.c | 3 +-- sound/pci/pcxhr/pcxhr_mix22.c | 2 +- sound/pci/rme9652/hdspm.c | 6 ++---- sound/soc/fsl/lpc3xxx-i2s.c | 3 +-- sound/soc/soc-dapm.c | 3 +-- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- sound/soc/sof/intel/hda-stream.c | 4 +--- sound/soc/sof/pcm.c | 2 +- sound/soc/stm/stm32_i2s.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 16 ++++++++-------- 12 files changed, 36 insertions(+), 43 deletions(-) -- 2.43.0

3 14

[PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Vijendar Mukunda 02 Aug '24

02 Aug '24

This patch series majorly consists of below changes. - Rename structures, macros and codec helper names used in Intel SoundWire generic driver to make it generic. - Move Intel SoundWire driver common structures, macros and codec helper functions to common placeholder so that it can be used by other platform machine driver. - Refactor few SoundWire common codec helper functions. - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. This work started a couple of months ago to avoid duplication of code that wasn't really Intel-specific in the "sof_sdw" machine driver. The code went through multiple iterations, was tested for multiple weeks and a couple of build issues reported by the Intel kbuild bots were corrected. This is the initial version of SoundWire machine driver for AMD platforms. Additional code refactoring will be done in the next step on the AMD side. Link: https://github.com/thesofproject/linux/pull/5068 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (31): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver include/sound/soc-acpi.h | 2 + include/sound/soc_sdw_utils.h | 216 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++ sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 9 - sound/soc/intel/boards/sof_sdw.c | 1142 ++--------------- sound/soc/intel/boards/sof_sdw_common.h | 166 +-- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../soc_sdw_dmic.c} | 10 +- .../soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../soc_sdw_rt700.c} | 12 +- .../soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 32 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 990 ++++++++++++++ sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + 38 files changed, 2475 insertions(+), 1357 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (87%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c -- 2.34.1

4 34

common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #413 was edited from ujfalusi: Hi, The PR will introduce a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

common: pcm/split: add support up to 32 / 8 channels
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #417 was edited from perexg: **DO NOT MERGE - after 1.2.12!** MOTU 828 has up to 32 hardware channels. Add support for this configuration to common/pcm/split.conf. Also, add support for up to 8 channels for the abstact devices (like 7.1 surround). Note that this change requires recent alsa-lib (Syntax 7). Link: https://github.com/alsa-project/alsa-ucm-conf/pull/416 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lag between microphone and speaker
by Guillermo Bernaldo de Quiros Maraver 02 Aug '24

02 Aug '24

Hi, good afternoon! First of all, I'm new to the mailing list so I'm not sure if this is the correct place to ask this question. I ask it here because I don't know a better place to ask it so, if this is not the correct place, please let me know. The thing is, I have a USB sound card plugged to a Beaglebone AI64. This works correctly (I mean, I can record my voice and I can play a wav file without problems. using the arecord and aplay tools) The problem comes with a software I'm writing which needs to read audio from the microphone, then do a process (which takes less than 20ms) and then, after that process, I have to write the audio to the speaker. The software, right now, consists of several files, but, the most important components of the software are Recorder and Player CPP classes which I show you at the end. The important part is the next: 1º I init the alsa context using the next calls: audioPlayer.OpenDevice("plughw:CARD=Device,DEV=0"); micRecorder.OpenDevice("plughw:CARD=Device,DEV=0"); 2º I'm using the sample rate of 16KHz, a sample size of 16bit and 1 channel. So, assuming I've done correctly, this should give me a total frame rate of 342 samples, then, if I multiply it by 2 bytes, I get 684 bytes of total size. 3º Finally, in the main, I do the next thing: while(1) { //bool keyPressed = id.IsKeyPressed(0); //printf("Key is pressed: %s\n", keyPressed ? "true" : "false"); const int8_t *pAudioBuffer = micRecorder.ReadAudio(); if(pAudioBuffer) { // Process a single frame of audio // Note: output_frame can be the same buffer as input_frame for in-place processing if(keyPressed) { // DO A PROCESS WHICH TAKES 19ms more or less } else { memcpy(buff_in,pAudioBuffer, buff_size); } // Opcionalmente, leemos/escribinos de/a fichero. //write(noisy_raw_fd, pAudioBuffer, micRecorder.GetBufferSize()); //write(clean_raw_fd, g_out_buffer, g_out_buffer_size); if(audioPlayer.WriteAudio((int8_t*)buff_in)<0) { return 1; } } //printf("Numero de frames escritos: %d del total de: %d\n", pcm, frames); } This works correctly, the problem is after some time I start to note that there is a difference of time between my speaking and my listening and I don't know why. I've enabled all traces in both the recorder and the player, but I don't see any error. Is just that there is a difference between my speaking and my listening. The difference is up to 10 seconds or more. Please, can anyone help me with this problem? PD: Please, be aware I'm just a software engineer, I'm not an expert in all technical stuff related to audio. I did this code based on some examples I found on internet which are: I used this code to implement the player class: https://gist.github.com/ghedo/963382/815c98d1ba0eda1b486eb9d80d9a91a81d9952… I used this code to implement the recorder class: https://gist.github.com/albanpeignier/104902 Here is the code // Player.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Player { public: Player(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Player(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); int WriteAudio(const int8_t *pAudioBuffer); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Player.cpp #include "Player.h" Player::Player(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Player::~Player() { CloseDevice(); } int Player::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } void Player::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); //free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } size_t Player::GetFrameSize() { return m_FrameBufferSize; } size_t Player::GetBufferSize() { return m_BufferSize; } int Player::WriteAudio(const int8_t *pAudioBuffer) { int pcm; if (pcm = snd_pcm_writei(m_AudioHandle, pAudioBuffer, FRAME_SIZE) == -EPIPE) { printf("XRUN.\n"); snd_pcm_prepare(m_AudioHandle); } else if (pcm < 0) { printf("ERROR. Can't write to PCM device. %s\n", snd_strerror(pcm)); return 0; } return 1; } int Player::InitAudioContext() { std::cout << "Iniciando Player" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "format width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(SND_PCM_FORMAT_S16_LE) / 8 * CHANNELS; return 0; } // Recorder.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Recorder { public: Recorder(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Recorder(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); const int8_t *ReadAudio(); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Recorder.cpp #include "Recorder.h" /* * Recuperacion ante errores de alsa. */ static int xrun_recovery(snd_pcm_t *handle, int err) { if (err == -EPIPE) { /* under-run */ printf("underrun!\n"); err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } Recorder::Recorder(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Recorder::~Recorder() { CloseDevice(); } int Recorder::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } size_t Recorder::GetFrameSize() { return m_FrameBufferSize; } /* * Tamaño total en bytes del buffer de memoria reservado */ size_t Recorder::GetBufferSize() { return m_BufferSize; } /* * Lee en el puntero de memoria resevado, las muestras de audio leidas de la capturadora de audio (microfono). */ const int8_t *Recorder::ReadAudio() { int err; if ((err = snd_pcm_readi(m_AudioHandle, m_pAudioBuffer, m_FrameBufferSize)) != m_FrameBufferSize) { // Si el error es de tipo PIPE o STRPIPE, tratamos de recuperar el dispositivo. if(xrun_recovery(m_AudioHandle, err) < 0) { std::cerr << "read from audio interface failed " << snd_strerror(err) << std::endl; return nullptr; } } return m_pAudioBuffer; } void Recorder::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } int Recorder::InitAudioContext() { std::cout << "Iniciando Grabador" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "formt width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(format) / 8 * CHANNELS; // Puntero al buffer de memoria reservado. m_pAudioBuffer = (int8_t *)malloc(m_BufferSize); return 0; }

1 0

[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 01 Aug '24

01 Aug '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 24ae1d4959be..1c6e035fd313 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static const struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.46.0.rc2.264.g509ed76dc8-goog

3 2

[PATCH] slimbus: generate MODULE_ALIAS() from MODULE_DEVICE_TABLE()
by Masahiro Yamada 01 Aug '24

01 Aug '24

Commit 9e663f4811c6 ("slimbus: core: add support to uevent") added the MODALIAS=slim:* uevent variable, but modpost does not generate the corresponding MODULE_ALIAS(). To support automatic module loading, slimbus drivers still need to manually add MODULE_ALIAS("slim:<manf_id>:<prod_code>:*"), as seen in sound/soc/codecs/wcd9335.c. To automate this, make modpost generate the proper MODULE_ALIAS() from MODULE_DEVICE_TABLE(slim, ). Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- scripts/mod/devicetable-offsets.c | 4 ++++ scripts/mod/file2alias.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index 518200813d4e..9c7b404defbd 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -153,6 +153,10 @@ int main(void) DEVID_FIELD(i3c_device_id, part_id); DEVID_FIELD(i3c_device_id, extra_info); + DEVID(slim_device_id); + DEVID_FIELD(slim_device_id, manf_id); + DEVID_FIELD(slim_device_id, prod_code); + DEVID(spi_device_id); DEVID_FIELD(spi_device_id, name); diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 5d1c61fa5a55..99dce93a4188 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -960,6 +960,16 @@ static int do_i3c_entry(const char *filename, void *symval, return 1; } +static int do_slim_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD(symval, slim_device_id, manf_id); + DEF_FIELD(symval, slim_device_id, prod_code); + + sprintf(alias, "slim:%x:%x:*", manf_id, prod_code); + + return 1; +} + /* Looks like: spi:S */ static int do_spi_entry(const char *filename, void *symval, char *alias) @@ -1555,6 +1565,7 @@ static const struct devtable devtable[] = { {"rpmsg", SIZE_rpmsg_device_id, do_rpmsg_entry}, {"i2c", SIZE_i2c_device_id, do_i2c_entry}, {"i3c", SIZE_i3c_device_id, do_i3c_entry}, + {"slim", SIZE_slim_device_id, do_slim_entry}, {"spi", SIZE_spi_device_id, do_spi_entry}, {"dmi", SIZE_dmi_system_id, do_dmi_entry}, {"platform", SIZE_platform_device_id, do_platform_entry}, -- 2.43.0

2 2

[PATCH 0/3] slimbus: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 01 Aug '24

01 Aug '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (3): slimbus: messaging: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ctrl: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ngd-ctrl: use 'time_left' variable with wait_for_completion_timeout() drivers/slimbus/messaging.c | 9 +++++---- drivers/slimbus/qcom-ctrl.c | 7 ++++--- drivers/slimbus/qcom-ngd-ctrl.c | 29 ++++++++++++++++------------- 3 files changed, 25 insertions(+), 20 deletions(-) -- 2.43.0

3 6

Re: [PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Mukunda,Vijendar 01 Aug '24

01 Aug '24

On 01/08/24 16:27, Charles Keepax wrote: > On Thu, Aug 01, 2024 at 02:44:15PM +0530, Vijendar Mukunda wrote: >> This patch series majorly consists of below changes. >> - Rename structures, macros and codec helper names used in Intel >> SoundWire generic driver to make it generic. >> - Move Intel SoundWire driver common structures, macros and codec >> helper functions to common placeholder so that it can be used by >> other platform machine driver. >> - Refactor few SoundWire common codec helper functions. >> - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. >> >> This work started a couple of months ago to avoid duplication of code >> that wasn't really Intel-specific in the "sof_sdw" machine driver. >> The code went through multiple iterations, was tested for multiple weeks >> and a couple of build issues reported by the Intel kbuild bots were >> corrected. >> >> This is the initial version of SoundWire machine driver for AMD >> platforms. Additional code refactoring will be done in the next step on >> the AMD side. >> >> Link: https://github.com/thesofproject/linux/pull/5068 >> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> >> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> >> > Would be good to also include the linux-sound mailing list on the > cover letter, if not the individuals given they are basically the > same on each patch. > > Also assuming your mail timed out again, you should be able to do > a separate git send-email for the later patches using the > --in-reply-to option to give the Message-ID of the cover letter > and they should append to the same chain. > > Thanks, > Charles Thanks, Charles, for your suggestion. I have shared the rest of the Patch series. - Vijendar

1 0

[PATCH 00/31] AMD SOF based generic SoundWire machine driver
by Vijendar Mukunda 01 Aug '24

01 Aug '24

This patch series majorly consists of below changes. - Rename structures, macros and codec helper names used in Intel SoundWire generic driver to make it generic. - Move Intel SoundWire driver common structures, macros and codec helper functions to common placeholder so that it can be used by other platform machine driver. - Refactor few SoundWire common codec helper functions. - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant Link: https://github.com/thesofproject/linux/pull/5068 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (31): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver include/sound/soc-acpi.h | 2 + include/sound/soc_sdw_utils.h | 216 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++ sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 9 - sound/soc/intel/boards/sof_sdw.c | 1142 ++--------------- sound/soc/intel/boards/sof_sdw_common.h | 166 +-- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../soc_sdw_dmic.c} | 10 +- .../soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../soc_sdw_rt700.c} | 12 +- .../soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 32 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 990 ++++++++++++++ sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + 38 files changed, 2475 insertions(+), 1357 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (87%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c -- 2.34.1

3 24

gitcompile: restore ACLOCAL_FLAGS ability to use ../alsa-lib/utils/
by GitHub pull_request - edited 01 Aug '24

01 Aug '24

alsa-project/alsa-utils pull request #272 was edited from marc-hb: Fixes cryptic "error macro 'AM_PATH_ALSA' not found in library configure.ac:22" when attempting a local install as described at https://thesofproject.github.io/latest/getting_started/build-guide/build-fr… Fixes giant commit f9e6010d5ec7 ("topology: plugins - add Intel nhlt encoder plugin") which added a second invocation of `aclocal` in 2021 but forgot the extra `$alsa_m4_flags`. To avoid duplication, do not fix by adding the missing `$alsa_m4_flags` to the second aclocal invocation. Instead, drop that variable and append `-I ../alsa-lib/utils` directly to $ACLOCAL_FLAGS Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/272 Patch URL : https://github.com/alsa-project/alsa-utils/pull/272.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

gitcompile: restore ACLOCAL_FLAGS ability to use ../alsa-lib/utils/
by GitHub pull_request - opened 01 Aug '24

01 Aug '24

alsa-project/alsa-utils pull request #272 was opened from marc-hb: Fixes cryptic "error macro 'AM_PATH_ALSA' not found in library configure.ac:22" when attempting a local install as described at https://thesofproject.github.io/latest/getting_started/build-guide/build-fr… Fixes giant commit f9e6010d5ec7 ("topology: plugins - add Intel nhlt encoder plugin") which added a second invocation of `aclocal` in 2021 but forgot the extra `$alsa_m4_flags`. To avoid duplication, do not fix by adding the missing `$alsa_m4_flags` to the second aclocal invocation. Instead, drop that variable and append `-I ../alsa-lib/utils` directly to $ACLOCAL_FLAGS Request URL : https://github.com/alsa-project/alsa-utils/pull/272 Patch URL : https://github.com/alsa-project/alsa-utils/pull/272.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 31 Jul '24

31 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- Changes in v2: - v1 was accidentally the backport version for older kernels instead of the one for v6.11 sound/pci/hda/cs35l56_hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..f2f623f713d1 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -613,7 +617,7 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - else + else if (expose_dsp_controls) add_dsp_controls_required = true; cs35l56->base.fw_patched = false; -- 2.43.0

1 1

[PATCH][LINUX][ALSA][USB AUDIO CLASS 1] wrong channel ids for surround
by Sylvain BERTRAND 31 Jul '24

31 Jul '24

Hi, In the USB audio class 1 specifications, page 33 (3.7.2.3 Audio Channel Cluster Format): Surround is spatially defined as [rear], hence left surround should be SNDRV_CHMAP_RL and not SND_CHMAP_SL (Side). Same for right surround channel. Right? I am not a mailing list subscriber, CC me please. regards, -- Sylvain BERTRAND --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -244,8 +244,8 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, SNDRV_CHMAP_FR, /* right front */ SNDRV_CHMAP_FC, /* center front */ SNDRV_CHMAP_LFE, /* LFE */ - SNDRV_CHMAP_SL, /* left surround */ - SNDRV_CHMAP_SR, /* right surround */ + SNDRV_CHMAP_RL, /* left surround */ + SNDRV_CHMAP_RR, /* right surround */ SNDRV_CHMAP_FLC, /* left of center */ SNDRV_CHMAP_FRC, /* right of center */ SNDRV_CHMAP_RC, /* surround */

2 3

[PATCH -next] ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate'
by Yue Haibing 31 Jul '24

31 Jul '24

The variable savedbitclkrate is assigned and never used, so can be removed. sound/soc/fsl/lpc3xxx-i2s.c:42:13: warning: variable ‘savedbitclkrate’ set but not used [-Wunused-but-set-variable] Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index af995ca081a3..62ef624d6dd4 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -39,7 +39,7 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 { u32 i2srate; u32 idxx, idyy; - u32 savedbitclkrate, diff, trate, baseclk; + u32 diff, trate, baseclk; /* Adjust rate for sample size (bits) and 2 channels and offset for * divider in clock output @@ -53,14 +53,12 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 /* Find the best divider */ *clkx = *clky = 0; - savedbitclkrate = 0; diff = ~0; for (idxx = 1; idxx < 0xFF; idxx++) { for (idyy = 1; idyy < 0xFF; idyy++) { trate = (baseclk * idxx) / idyy; if (abs(trate - i2srate) < diff) { diff = abs(trate - i2srate); - savedbitclkrate = trate; *clkx = idxx; *clky = idyy; } -- 2.34.1

3 2

[PATCH] ASoC: rsnd: use pcm_dmaengine code
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd is implementing own DMAEngine code, but we can replace it with pcm_dmaengine code, because these are almost same. Let's use existing and stable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 17 --------- sound/soc/sh/rcar/dma.c | 75 ++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 2 +- 5 files changed, 9 insertions(+), 87 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7bddfd5e38d6c..426632996a0a3 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_RCAR depends on COMMON_CLK depends on OF select SND_SIMPLE_CARD_UTILS + select SND_DMAENGINE_PCM select REGMAP_MMIO help This option enables R-Car SRU/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6bc7027ed4dbf..233128c8ff145 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -660,23 +660,6 @@ static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) return rsnd_rdai_get(priv, dai->id); } -/* - * rsnd_soc_dai functions - */ -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io) -{ - struct snd_pcm_substream *substream = io->substream; - - /* - * this function should be called... - * - * - if rsnd_dai_pointer_update() returns true - * - without spin lock - */ - - snd_pcm_period_elapsed(substream); -} - static void rsnd_dai_stream_init(struct rsnd_dai_stream *io, struct snd_pcm_substream *substream) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7b499eee50806..2342bbb6fe92e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -7,6 +7,7 @@ #include <linux/delay.h> #include <linux/of_dma.h> +#include <sound/dmaengine_pcm.h> #include "rsnd.h" /* @@ -22,8 +23,6 @@ struct rsnd_dmaen { struct dma_chan *chan; - dma_cookie_t cookie; - unsigned int dma_len; }; struct rsnd_dmapp { @@ -66,20 +65,6 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -static void __rsnd_dmaen_complete(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - if (rsnd_io_is_working(io)) - rsnd_dai_period_elapsed(io); -} - -static void rsnd_dmaen_complete(void *data) -{ - struct rsnd_mod *mod = data; - - rsnd_mod_interrupt(mod, __rsnd_dmaen_complete); -} - static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_dai_stream *io, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) @@ -98,13 +83,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - - if (dmaen->chan) - dmaengine_terminate_async(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_STOP); } static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, @@ -120,7 +99,7 @@ static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, * Let's call it under prepare */ if (dmaen->chan) - dma_release_channel(dmaen->chan); + snd_dmaengine_pcm_close_release_chan(io->substream); dmaen->chan = NULL; @@ -153,7 +132,7 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, return -EIO; } - return 0; + return snd_dmaengine_pcm_open(io->substream, dmaen->chan); } static int rsnd_dmaen_start(struct rsnd_mod *mod, @@ -162,12 +141,9 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - int is_play = rsnd_io_is_play(io); int ret; /* @@ -195,7 +171,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, } } - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.direction = snd_pcm_substream_to_dma_direction(io->substream); cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = buswidth; @@ -209,32 +185,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return -EIO; - } - - desc->callback = rsnd_dmaen_complete; - desc->callback_param = rsnd_mod_get(dma); - - dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); - - dmaen->cookie = dmaengine_submit(desc); - if (dmaen->cookie < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return -EIO; - } - - dma_async_issue_pending(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_START); } struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, @@ -307,19 +258,7 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct dma_tx_state state; - enum dma_status status; - unsigned int pos = 0; - - status = dmaengine_tx_status(dmaen->chan, dmaen->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - if (state.residue > 0 && state.residue <= dmaen->dma_len) - pos = dmaen->dma_len - state.residue; - } - *pointer = bytes_to_frames(runtime, pos); + *pointer = snd_dmaengine_pcm_pointer(io->substream); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..dba684e4334b1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -576,7 +576,6 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, #define rsnd_rdai_width_get(rdai) \ rsnd_rdai_width_ctrl(rdai, 0) int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8d2a86383ae01..b3d4e8ae07eff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -706,7 +706,7 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, spin_unlock(&priv->lock); if (elapsed) - rsnd_dai_period_elapsed(io); + snd_pcm_period_elapsed(io->substream); if (stop) snd_pcm_stop_xrun(io->substream); -- 2.43.0

2 1

[PATCH] ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd_mod_confirm_ssi() confirms mod sanity, it should always be confirmed, not only when DEBUG. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index afd69c6eb6544..0f190abf00e75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -262,7 +262,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; - rsnd_mod_confirm_src(src_mod); + rsnd_mod_make_sure(src_mod, RSND_MOD_SRC); rsnd_adg_get_timesel_ratio(priv, io, in_rate, out_rate, @@ -291,7 +291,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int shift = (id % 4) * 8; u32 mask = 0xFF << shift; - rsnd_mod_confirm_ssi(ssi_mod); + rsnd_mod_make_sure(ssi_mod, RSND_MOD_SSI); val = val << shift; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..8cf5d9001f437 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -871,15 +871,6 @@ void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); -#ifdef DEBUG -#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) -#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) -#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -#else -#define rsnd_mod_confirm_ssi(mssi) -#define rsnd_mod_confirm_src(msrc) -#define rsnd_mod_confirm_dvc(mdvc) -#endif /* * If you don't need interrupt status debug message, -- 2.43.0

2 1

[PATCH 0/2] ASoC: fsl_micfil: Check the difference for i.MX8 and i.MX9
by Shengjiu Wang 31 Jul '24

31 Jul '24

There are some register difference for i.MX8 and i.MX9 REG_MICFIL_FIFO_CTRL definition is updated. REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID, REG_MICFIL_PARAM are added from i.MX9 Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms sound/soc/fsl/fsl_micfil.c | 20 +++++++++++++++----- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 16 insertions(+), 6 deletions(-) -- 2.34.1

3 4

Re: [PATCH v1] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Jaroslav Kysela 31 Jul '24

31 Jul '24

On 30. 07. 24 16:37, Stefan Binding wrote: > Add a kernel parameter to allow coefficients to be exposed as ALSA controls. > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls by default. > > However, nothing in userspace currently uses them, and is unlikely to > do so in the future, therefore we don't need to create ASLA controls > for them. > > These controls can be useful for debug, so we can add a kernel > parameter to re-enable them if necessary. > > Disabling these controls would prevent userspace from trying to read > these controls when the CS35L41 is hibernating, which ordinarily > would result in an error message. This is probably not a right argument to add this code. The codec should be powered up when those controls are accessed or those controls should be cached by the driver. Although the controls have not been used yet, exposing them in this way is not ideal. Could you fix the driver (no I/O errors)? Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 3

[PATCH alsa-lib 0/4] Add API helper functions for creating UMP Endpoint and Blocks
by Takashi Iwai 31 Jul '24

31 Jul '24

Hi, this is a patch set to alsa-lib to add the new API functions for creating a virtual UMP Endpoint and UMP Blocks arbitrarily from a user-space, that is, to create a virtual UMP device. An example program is included in test directory. Takashi === Takashi Iwai (4): ump: Add missing *_set variants for snd_ump_endpoint_info and snd_ump_block_info seq: Add API helper functions for creating UMP Endpoint and Blocks test: Add an example program to create a virtual UMP Endpoint Add test/seq-ump-example to .gitignore .gitignore | 1 + include/seqmid.h | 7 + include/ump.h | 36 ++++- src/rawmidi/ump.c | 304 +++++++++++++++++++++++++++++++++++++++-- src/seq/seq.c | 6 +- src/seq/seq_local.h | 4 + src/seq/seqmid.c | 249 +++++++++++++++++++++++++++++++++ test/Makefile.am | 3 +- test/seq-ump-example.c | 187 +++++++++++++++++++++++++ 9 files changed, 783 insertions(+), 14 deletions(-) create mode 100644 test/seq-ump-example.c -- 2.43.0

2 8

[PATCH v4 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 31 Jul '24

31 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

3 4

Bug in tlv320adc3101 driver
by Terry Sanders 31 Jul '24

31 Jul '24

Hi, There is a bug in the code for the tlv320adc3101 driver when setting up the D value for PLL. The D value is set via I2C using page 0 registers 7 (MSB) and 8 (LSB). The data sheet has the following information for the registers: "Page 0 / Register 7 will be updated when Page 0 / Register 8 is written immediately after Page 0 / Register 7 is written" and: "Page 0 / Register 8 must be written immediately after writing to Page 0 / Register 7" The function that implements this is adc3xxx_setup_pll() in file soc/codecs/tlv320adc3xxx.c line 1172 - 1189. The function is shown below: static void adc3xxx_setup_pll(struct snd_soc_component *component, int div_entry) { int i = div_entry; /* P & R values */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); /* J value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); /* D value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); } In the function the LSB (register 8) is written first followed by the MSB (register 7). This is the wrong way around and should be swapped so that the MSB is written first. Regards Terry Sanders

2 1

[PATCH v1] ASoc: tas2781: Move tas2563_dvc_table into a separate Header file
by Shenghao Ding 30 Jul '24

30 Jul '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..cf6eedb98ad2 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

3 4

[PATCH v1] ASoc: tas2781: Add TAS2563 into the Header
by Shenghao Ding 30 Jul '24

30 Jul '24

Add TAS2563 into the Header in case of misunderstanding and add channel No information for error debug in tasdevice_dev_read. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header - Add channel no into the log in tasdevice_dev_read for error debug --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } -- 2.34.1

2 1

[PATCH v3 0/3] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 30 Jul '24

30 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (3): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" ALSA: firewire-lib: amdtp-stream work queue inline description sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

2 4

[PATCH] ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding
by Krzysztof Kozlowski 30 Jul '24

30 Jul '24

The APQ8016 SBC and MSM8916 QDSP6 sound cards are a bit different from others: they have additional IO muxing address space and pin control. Move them to separate schema, so the original qcom,sm8250.yaml will be easier to manage. New schema is going to grow for other platforms having more of IO muxing address spaces. Cc: Adam Skladowski <a39.skl(a)gmail.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++++++++++++++++++ .../bindings/sound/qcom,sm8250.yaml | 137 ------------ 2 files changed, 205 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000000..6ad451549036 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index c9076dcd44c1..1d3acdc0c733 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -27,9 +27,7 @@ properties: - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - - qcom,apq8016-sbc-sndcard - qcom,apq8096-sndcard - - qcom,msm8916-qdsp6-sndcard - qcom,qcm6490-idp-sndcard - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard @@ -58,18 +56,6 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User visible long sound card name - pin-switches: - description: List of widget names for which pin switches should be created. - $ref: /schemas/types.yaml#/definitions/string-array - - widgets: - description: User specified audio sound widgets. - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - - # Only valid for some compatibles (see allOf if below) - reg: true - reg-names: true - patternProperties: ".*-dai-link$": description: @@ -122,34 +108,6 @@ required: - compatible - model -allOf: - - if: - properties: - compatible: - contains: - enum: - - qcom,apq8016-sbc-sndcard - - qcom,msm8916-qdsp6-sndcard - then: - properties: - reg: - items: - - description: Microphone I/O mux register address - - description: Speaker I/O mux register address - reg-names: - items: - - const: mic-iomux - - const: spkr-iomux - required: - - compatible - - model - - reg - - reg-names - else: - properties: - reg: false - reg-names: false - additionalProperties: false examples: @@ -231,98 +189,3 @@ examples: }; }; }; - - - | - #include <dt-bindings/sound/qcom,lpass.h> - sound@7702000 { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "DB410c"; - audio-routing = - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1"; - - pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; - pinctrl-names = "default", "sleep"; - - quaternary-dai-link { - link-name = "ADV7533"; - cpu { - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; - - primary-dai-link { - link-name = "WCD"; - cpu { - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - tertiary-dai-link { - link-name = "WCD-Capture"; - cpu { - sound-dai = <&lpass MI2S_TERTIARY>; - }; - codec { - sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; - }; - }; - }; - - - | - #include <dt-bindings/sound/qcom,q6afe.h> - #include <dt-bindings/sound/qcom,q6asm.h> - sound@7702000 { - compatible = "qcom,msm8916-qdsp6-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "msm8916"; - widgets = - "Speaker", "Speaker", - "Headphone", "Headphones"; - pin-switches = "Speaker"; - audio-routing = - "Speaker", "Speaker Amp OUT", - "Speaker Amp IN", "HPH_R", - "Headphones", "HPH_L", - "Headphones", "HPH_R", - "AMIC1", "MIC BIAS Internal1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS Internal3"; - aux-devs = <&speaker_amp>; - - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&cdc_pdm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - primary-dai-link { - link-name = "Primary MI2S"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - platform { - sound-dai = <&q6routing>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - }; -- 2.43.0

6 6

[PATCH 0/6] ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports mask
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Incorrect mask of Soundwire ports - one bit too long/big - was passed. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Not adding Cc-stable because I don't think the issue can lead to real out-of-bounds read. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask sound/soc/codecs/wcd937x-sdw.c | 4 ++-- sound/soc/codecs/wcd938x-sdw.c | 4 ++-- sound/soc/codecs/wcd939x-sdw.c | 4 ++-- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- 6 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240726-asoc-wcd-wsa-swr-ports-genmask-868239f9682f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 8

[PATCH] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 29 Jul '24

29 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..1494383b22c9 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -758,6 +762,9 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas cs35l56_hda_create_controls(cs35l56); + if (expose_dsp_controls) + cs35l56_hda_add_dsp_controls(cs35l56); + #if IS_ENABLED(CONFIG_SND_DEBUG) cs35l56->debugfs_root = debugfs_create_dir(dev_name(cs35l56->base.dev), sound_debugfs_root); cs_dsp_init_debugfs(&cs35l56->cs_dsp, cs35l56->debugfs_root); -- 2.43.0

2 2

ASUS RoG Strix Z790-F: SPDIF (PCM) has wrong channels | how to get passthrough working?
by GitHub issues - opened 29 Jul '24

29 Jul '24

alsa-project/alsa-ucm-conf issue #434 was opened from Nightwing0815: Hi there, finally, [after a long journey,](https://github.com/alsa-project/alsa-ucm-conf/issues/350) yesterday I did an update from linuxmint to newest version 22, based on ubuntu 24.04 (I'll attach alsa-info and a system info). Now it's no big problem any more, to get SPDIF working: I just selected the right device and muted all others in alsamixer, except the pcm signal. But hey, there's a new problem: With a speaker-test -c6, I'm not able to hear my rear channels and the lfe channel, where they should be. Rear coming on front speakers... So, what do I have to do? Further, I'm asking for help to configure a pcm signal passthru for dts and (e)ac3 soundfiles. My teufel receiver recognices only a 5.1 stereo source, not an DD5.1 or DTS signal... [alsa-info.txt](https://github.com/user-attachments/files/16417700/alsa-info… [sysinfo.txt](https://github.com/user-attachments/files/16417724/sysinfo.txt) Thanks in advance for getting this problems solved, ToM Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/434 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/7] ASoC: codecs: wsa88xx and wcd93xx: Soundwire port non-functional cleanup
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups to make the code more robust or readable. No functional impact (compiled objects stay the same). Best regards, Krzysztof --- Krzysztof Kozlowski (7): ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- sound/soc/codecs/wcd938x.c | 18 ------------------ sound/soc/codecs/wcd938x.h | 4 ++-- sound/soc/codecs/wcd939x.h | 6 +++--- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 7 files changed, 88 insertions(+), 92 deletions(-) --- base-commit: 66df68b9536e7465aec703733ea91ca39b787ed8 change-id: 20240725-asoc-wsa88xx-port-arrays-c58c96cd888e Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH v2] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 29 Jul '24

29 Jul '24

This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Changes in v2: - Update changelog - Link to v1: https://lore.kernel.org/r/20240721-sti-audio-fix-v1-1-a8b1ecf61cb4@free.fr --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

2 1

[RESEND] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 29 Jul '24

29 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

2 1

[PATCH] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8351f0974e6a..4158657914fb 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2296,36 +2296,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); -- 2.43.0

2 1

[PATCH] ASoC: cs42l42: Convert comma to semicolon
by Chen Ni 29 Jul '24

29 Jul '24

Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; -- 2.25.1

4 3

[PATCH 0/4] ASoC: codecs: wsa88xx: Few cleanups
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups around wsa88xx codecs. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant sound/soc/codecs/wsa881x.c | 2 -- sound/soc/codecs/wsa883x.c | 33 ++++++++++++++++++--------------- sound/soc/codecs/wsa884x.c | 5 ++--- 3 files changed, 20 insertions(+), 20 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240710-asoc-wsa88xx-version-dd0039265245 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 5

Re: [PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by edmund.raile 29 Jul '24

29 Jul '24

> Thank you for your sending the revised patches, it looks better than the > previous one. However, I have an additional request. Allright, patch v3 it is. > [1] https://git-scm.com/docs/git-revert Should have known git has something like that, how handy! > $ git revert -s b5b519965c4c Yes, 5b5 can be removed via revert, but what is the difference in effect? Just time saving? > $ git revert -s 7ba5ca32fe6e This one I'd like to ask you about: The original inline comment in amdtp-stream.c amdtp_domain_stream_pcm_pointer() ``` // This function is called in software IRQ context of // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. // Therefore, no need to flush the queue in buffer furthermore. // // When the process context reach here, some packets will be // already queued in the buffer. These packets should be handled // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above ``` (let's call the above v1) was replaced with ``` // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. ``` on occasion of 7ba5ca32fe6e (let's call this v2). I sought to replace it with ``` // use wq to prevent deadlock between process context spin_lock // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and // softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() // in snd_pcm_period_elapsed() ``` to prevent this issue from occurring again (let's call this v3). Should I include v1, v3 or a combination of v1 and v3 in my next patch? > Just for safe, it is preferable to execute 'scripts/checkpatch.pl' in > kernel tree to check the patchset generated by send-email subcommand[3]. Absolutely should have done so, sorry. Thank you for your patience and guidance, Edmund Raile.

2 1

[PATCH AUTOSEL 5.15 09/13] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 089b6c7994f9a..cfa0c3eaffea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 10/15] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cf501baf6f143..9cd85ab19c553 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -387,6 +387,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/15] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d03de37e3578c..cf501baf6f143 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -267,6 +267,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 11/17] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index dc144cd7e0e3b..db1dcb9d70466 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -425,6 +425,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 10/17] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5980fce817976..dc144cd7e0e3b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -286,6 +286,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 15/23] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 965d46968f95a..0b3b2c88d31ed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -406,6 +406,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 14/23] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a878b6b3d1948..965d46968f95a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -277,6 +277,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 02/23] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 28 Jul '24

28 Jul '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e41b0d95e0ff7..a878b6b3d1948 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -505,6 +505,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

1 0

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/4] MSM8953/MSM8976 ASoC support
by Adam Skladowski 28 Jul '24

28 Jul '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Adam Skladowski (1): ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard Vladimir Lypak (3): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add cajon and cajon v2 support ASoC: qcom: apq8016_sbc: Add support for msm8953/msm8976 SoC .../sound/qcom,apq8016-sbc-sndcard.yaml | 51 +++++++++-- sound/soc/codecs/msm8916-wcd-analog.c | 63 +++++++++++++- sound/soc/qcom/apq8016_sbc.c | 84 ++++++++++++++++++- 3 files changed, 186 insertions(+), 12 deletions(-) -- 2.45.2

3 9

[PATCH v23 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 26 Jul '24

26 Jul '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Wesley Cheng (32): ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/xhci-plat.c | 2 + .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 188 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 5705 insertions(+), 29 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

4 59

[PATCH 0/2] {ASoC,media}: constify snd_soc_component_driver struct
by Javier Carrasco 26 Jul '24

26 Jul '24

The `snd_soc_component_driver` struct is never modified after its declaration, and its only direct user `devm_snd_soc_register_component()` expects a constant value anyway. Declare `snd_soc_component_driver` as const to move their declarations to read-only sections. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

3 6

[GIT PULL] ASoC fixes for v6.11-merge-window
by Mark Brown 25 Jul '24

25 Jul '24

The following changes since commit c51cba4755609ad97ba97713210c16f043c73224: Fix the unbalanced pm_runtime_enable in wcd937x-sdw (2024-07-12 17:38:18 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-merge-window for you to fetch changes up to ab53dfdcdd1ec8df8729890aefa5b0e3c900afbb: ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components (2024-07-25 14:20:30 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A selection of routine fixes and quirks that came in since the merge window. The fsl-asoc-card change is a fix for systems with multiple cards where updating templates in place leaks data from one card to another. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: tegra: select CONFIG_SND_SIMPLE_CARD_UTILS Curtis Malainey (1): ASoC: Intel: Fix RT5650 SSP lookup Dan Carpenter (1): ASoC: TAS2781: Fix tasdev_load_calibrated_data() Daniel Baluta (1): ASoC: SOF: imx8m: Fix DSP control regmap retrieval Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Only handle dai_config with HW_PARAMS for ChainDMA ASoC: SOF: ipc4-topology: Preserve the DMA Link ID for ChainDMA on unprepare Pierre-Louis Bossart (2): ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable Shengjiu Wang (1): ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components Takashi Iwai (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E16 Gen 2 Venkata Prasad Potturu (1): ASoC: sof: amd: fix for firmware reload failure in Vangogh platform sound/soc/amd/yc/acp6x-mach.c | 7 ++++ sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++-------- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 9 +++++ sound/soc/intel/common/soc-intel-quirks.h | 2 +- sound/soc/sof/amd/pci-vangogh.c | 1 - sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/hda-loader.c | 20 ++++++---- sound/soc/sof/intel/hda.c | 17 ++++---- sound/soc/sof/ipc4-topology.c | 18 +++++++-- sound/soc/tegra/Kconfig | 1 + 11 files changed, 85 insertions(+), 40 deletions(-)

2 1

[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 25 Jul '24

25 Jul '24

Bisection revealed that the bitcrushing distortion with RME FireFace 800 was caused by 1d717123bb1a7555 ("ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning"). Reverting this commit yields restoration of clear audio output. I will send in a patch reverting this commit for now, soonTM. #regzbot introduced: 1d717123bb1a7555

4 6

[PATCH] ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components
by Shengjiu Wang 25 Jul '24

25 Jul '24

The static snd_soc_dai_link_components cause conflict for multiple instances of this generic driver. For example, when there is wm8962 and SPDIF case enabled together, the contaminated snd_soc_dai_link_components will cause another device probe fail. Fixes: 6d174cc4f224 ("ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++++++++++------------- 1 file changed, 29 insertions(+), 17 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 82df887b3af5..f6c3aeff0d8e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -306,27 +306,12 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_fe, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_be, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); - static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", .stream_name = "HiFi", .ops = &fsl_asoc_card_ops, - SND_SOC_DAILINK_REG(hifi), }, /* DPCM Link between Front-End and Back-End (Optional) */ { @@ -335,7 +320,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .dynamic = 1, - SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-BE", @@ -345,7 +329,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(hifi_be), }, }; @@ -637,6 +620,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev[2] = { NULL, NULL }; + struct snd_soc_dai_link_component *dlc; const char *codec_dai_name[2]; const char *codec_dev_name[2]; u32 asrc_fmt = 0; @@ -717,7 +701,35 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + /* + * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms + * "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms + * "DPCM Link Back-End": 1 cpus, 2 codecs + * totally 10 components + */ + dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + ret = -ENOMEM; + goto asrc_fail; + } + + priv->dai_link[0].cpus = &dlc[0]; + priv->dai_link[0].num_cpus = 1; + priv->dai_link[0].codecs = &dlc[1]; priv->dai_link[0].num_codecs = 1; + priv->dai_link[0].platforms = &dlc[3]; + priv->dai_link[0].num_platforms = 1; + + priv->dai_link[1].cpus = &dlc[4]; + priv->dai_link[1].num_cpus = 1; + priv->dai_link[1].codecs = &dlc[5]; + priv->dai_link[1].num_codecs = 0; /* dummy */ + priv->dai_link[1].platforms = &dlc[6]; + priv->dai_link[1].num_platforms = 1; + + priv->dai_link[2].cpus = &dlc[7]; + priv->dai_link[2].num_cpus = 1; + priv->dai_link[2].codecs = &dlc[8]; priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; -- 2.34.1

2 1

[PATCH v5 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 25 Jul '24

25 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 124 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 593 insertions(+), 73 deletions(-) base-commit: 472438c7e0e2261c6737a8321f46ef176eef1c8f -- 2.25.1

3 3

[RFC v2 00/xx] ALSA: ALSA: add snd_pcm_is_playback/capture()
by Kuninori Morimoto 25 Jul '24

25 Jul '24

Hi Iwai-san, Mark Cc Sakamoto-san, Amadeusz, Pierre-Louis It seems we can use _Generic() more simply, v2 RFC is using it. I'm not 100% sure but unfortunately we can't use bit-field with _Generic() in gcc (clang seems possible to handle it ?). pci/ac97 is the only user of bit-field direction. To avoid posting patch-bomb, I will post main patch and some sample patches as RFC. Please review it. Link: https://lore.kernel.org/r/87zfqel1g7.wl-kuninori.morimoto.gx@renesas.com Thank you for your help !! Best regards --- Kuninori Morimoto

4 9

[kernel][usb audio class 1] surround channels mapped to wrong alsa channel ids
by GitHub issues - opened 25 Jul '24

25 Jul '24

alsa-project/alsa-lib issue #403 was opened from sylware: In $linux_src/sound/usb/stream.c, function "convert_chmap", in the "uac1_maps" array, the "left surround" should be SNDRV_CHMAP_RL instead of SNDRV_CHMAP_SL, and the "right surround" should be SNDRV_CHMAP_RR instead of SNDDRV_CHMAP_SR. Issue URL : https://github.com/alsa-project/alsa-lib/issues/403 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 24 Jul '24

24 Jul '24

Commit b5b519965c4c ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. With RME Fireface 800, this lead to a regression since Kernels 5.14.0, causing a deadlock with eventual system freeze under ALSA operation: ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci Restore the work queue to prevent deadlock between ALSA substream process context spin_lock of snd_pcm_stream_lock_irq() in snd_pcm_status64() and OHCI 1394 IT softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed(). to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Closes: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… Fixes: 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") Signed-off-by: Edmund Raile <edmund.raile(a)proton.me> --- This is the follow-up patch to the 5.14.0 regression I reported: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Takashi Sakamoto explained the issue in his response to the regression: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() This leads me to believe this isn't just an issue limited to the RME Fireface driver (snd_fireface), though I can not test the other devices. sound/firewire/amdtp-stream.c | 32 +++++++++++++++++++++----------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 22 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index d35d0a420ee0..77b99a2117f4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -77,6 +77,8 @@ // overrun. Actual device can skip more, then this module stops the packet streaming. #define IR_JUMBO_PAYLOAD_MAX_SKIP_CYCLES 5 +static void pcm_period_work(struct work_struct *work); + /** * amdtp_stream_init - initialize an AMDTP stream structure * @s: the AMDTP stream to initialize @@ -105,6 +107,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); + INIT_WORK(&s->period_work, pcm_period_work); s->packet_index = 0; init_waitqueue_head(&s->ready_wait); @@ -347,6 +350,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload); */ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) { + cancel_work_sync(&s->period_work); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; } @@ -611,19 +615,21 @@ static void update_pcm_pointers(struct amdtp_stream *s, // The program in user process should periodically check the status of intermediate // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. - if (!pcm->runtime->no_period_wakeup) { - if (in_softirq()) { - // In software IRQ context for 1394 OHCI. - snd_pcm_period_elapsed(pcm); - } else { - // In process context of ALSA PCM application under acquired lock of - // PCM substream. - snd_pcm_period_elapsed_under_stream_lock(pcm); - } - } + if (!pcm->runtime->no_period_wakeup) + queue_work(system_highpri_wq, &s->period_work); } } +static void pcm_period_work(struct work_struct *work) +{ + struct amdtp_stream *s = container_of(work, struct amdtp_stream, + period_work); + struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); + + if (pcm) + snd_pcm_period_elapsed(pcm); +} + static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params, bool sched_irq) { @@ -1854,7 +1860,10 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_softirq()) + // workqueue serves to prevent deadlock between process context spinlock + // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and softIRQ spinlock + // of snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() + if ((!in_softirq()) && (current_work() != &s->period_work)) fw_iso_context_flush_completions(irq_target->context); } @@ -1910,6 +1919,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s) return; } + cancel_work_sync(&s->period_work); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index a1ed2e80f91a..775db3fc4959 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -191,6 +191,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; + struct work_struct period_work; snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; unsigned int pcm_frame_multiplier; -- 2.45.2

2 1

[bug report] ASoC: Intel: cht_bsw_rt5645: Set card.components string
by Dan Carpenter 23 Jul '24

23 Jul '24

Hello Hans de Goede, Commit f87b4402163b ("ASoC: Intel: cht_bsw_rt5645: Set card.components string") from Nov 26, 2023 (linux-next), leads to the following Smatch static checker warning: sound/soc/intel/boards/cht_bsw_rt5645.c:587 snd_cht_mc_probe() error: we previously assumed 'adev' could be null (see line 581) sound/soc/intel/boards/cht_bsw_rt5645.c 570 /* set correct codec name */ 571 for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) 572 if (cht_dailink[i].codecs->name && 573 !strcmp(cht_dailink[i].codecs->name, 574 "i2c-10EC5645:00")) { 575 dai_index = i; 576 break; 577 } 578 579 /* fixup codec name based on HID */ 580 adev = acpi_dev_get_first_match_dev(mach->id, NULL, -1); 581 if (adev) { ^^^^ The old code assumes adev can be NULL 582 snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), 583 "i2c-%s", acpi_dev_name(adev)); 584 cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; 585 } 586 /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ --> 587 codec_dev = acpi_get_first_physical_node(adev); ^^^^ Unchecked dereference 588 acpi_dev_put(adev); 589 if (!codec_dev) 590 return -EPROBE_DEFER; 591 592 snd_soc_card_chtrt5645.components = rt5645_components(codec_dev); 593 snd_soc_card_chtrt5650.components = rt5645_components(codec_dev); 594 regards, dan carpenter

2 2

[PATCH for-6.10 0/2] ASoC: cs35l56: Set correct upper volume limit
by Richard Fitzgerald 23 Jul '24

23 Jul '24

Patch series to limit the upper range of the CS35L56 volume control to +12 dB. These commits were not marked 'Fixes' because they were thought to be only a cosmetic issue. The user could reduce the volume to a usable value. But for some complex audio topologies with SOF Audio DSP + CS42L43 + multiple CS35L56 it has turned out to be not obvious to the user what the problem actually is and what to do to fix it. As support for these topologies went into 6.10 we would like this series to be applied to 6.10. Richard Fitzgerald (2): ASoC: cs35l56: Use header defines for Speaker Volume control definition ASoC: cs35l56: Limit Speaker Volume to +12dB maximum include/sound/cs35l56.h | 2 +- sound/soc/codecs/cs35l56.c | 6 +++++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.39.2

2 3

[PATCH 0/4] MSM8953/MSM8976 ASoC support
by Adam Skladowski 23 Jul '24

23 Jul '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Adam Skladowski (1): ASoC: dt-bindings: qcom,sm8250: Add msm8953/msm8976-qdsp6-sndcard Vladimir Lypak (3): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add cajon and cajon v2 support ASoC: qcom: apq8016_sbc: Add support for msm8953/msm8976 SoC .../bindings/sound/qcom,sm8250.yaml | 28 ++++++- sound/soc/codecs/msm8916-wcd-analog.c | 63 +++++++++++++- sound/soc/qcom/apq8016_sbc.c | 82 ++++++++++++++++++- 3 files changed, 164 insertions(+), 9 deletions(-) -- 2.45.2

4 8

[RFC 00/xx] ALSA: ALSA: add snd_stream_is_playback/capture()
by Kuninori Morimoto 23 Jul '24

23 Jul '24

Hi Iwai-san, Mark Current many drivers are using below code to know its direction. if (direction == SNDRV_PCM_STREAM_PLAYBACK) I think it should be handled by function. But is it acceptable idea ? Because it will be too many patch-set, I want to know it was acceptable idea or not before posting patch-bomb. I will post main patch, and sample driver patches. Thank you for your help !! Best regards --- Kuninori Morimoto

5 17

[PATCH] ASoc: TAS2781: Fix tasdev_load_calibrated_data()
by Dan Carpenter 22 Jul '24

22 Jul '24

This function has a reversed if statement so it's either a no-op or it leads to a NULL dereference. Fixes: b195acf5266d ("ASoC: tas2781: Fix wrong loading calibrated data sequence") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 63626b982d04..8f9a3ae7153e 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2162,7 +2162,7 @@ static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) return; cal = cal_fmw->calibrations; - if (cal) + if (!cal) return; load_calib_data(priv, &cal->dev_data); -- 2.43.0

2 1

[PATCH] ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable
by Pierre-Louis Bossart 22 Jul '24

22 Jul '24

the Intel kbuild bot reports a link failure when IOSF_MBI is built-in but the Merrifield driver is configured as a module. The soc-intel-quirks.h is included for Merrifield platforms, but IOSF_MBI is not selected for that platform. ld.lld: error: undefined symbol: iosf_mbi_read >>> referenced by atom.c >>> sound/soc/sof/intel/atom.o:(atom_machine_select) in archive vmlinux.a This patch forces the use of the fallback static inline when IOSF_MBI is not reachable. Fixes: 536cfd2f375d ("ASoC: Intel: use common helpers to detect CPUs") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202407160704.zpdhJ8da-lkp@intel.com/ Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/common/soc-intel-quirks.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index de4e550c5b34..42bd51456b94 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,7 +11,7 @@ #include <linux/platform_data/x86/soc.h> -#if IS_ENABLED(CONFIG_X86) +#if IS_REACHABLE(CONFIG_IOSF_MBI) #include <linux/dmi.h> #include <asm/iosf_mbi.h> -- 2.43.0

2 1

[PATCH alsa-utils 00/10] Cleanup seq/* stuff and extend aseqsend
by Takashi Iwai 22 Jul '24

22 Jul '24

Hi, this is a series of cleanup / fix patches for alsa-utils/seq/* stuff, including the extension of aseqsend to support UMP packet bytes. Takashi === Takashi Iwai (10): configure: Requires the latest ALSA-lib release 1.2.12 aplaymidi: Drop ifdef for UMP support aplaymidi: Allow to pass 0 to -u option, too aconnect: Drop superfluous ifdefs aseqdump: Drop ifdef for UMP support aseqdump: Check the -u option value properly configure: Drop unused conditionals aseqdump: Show SysEx prefix to UMP SysEx data dump aseqsend: Support UMP mode aseqsend: Support long options configure.ac | 13 +------- seq/aconnect/aconnect.c | 31 ------------------ seq/aplaymidi/aplaymidi.c | 32 +++--------------- seq/aseqdump/aseqdump.c | 25 +++----------- seq/aseqsend/aseqsend.1 | 69 +++++++++++++++++++++------------------ seq/aseqsend/aseqsend.c | 54 +++++++++++++++++++++++++++++- 6 files changed, 100 insertions(+), 124 deletions(-) -- 2.43.0

1 10

[PATCH v3] ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
by wangdicheng 22 Jul '24

22 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> Sonix HD USB Camera does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. (snip) [1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd [1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00 [1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [1.984127] usb 3-3: Product: USB 2.0 Camera [1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd. [5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v2 -> v3: Update the insertion order of ID 0c45 v1 -> v2: Arrange the ID in order --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 58156fbca02c..b42654033c33 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2125,6 +2125,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0b0e, 0x0349, /* Jabra 550a */ QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ -- 2.25.1

2 1

[PATCH] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 22 Jul '24

22 Jul '24

Restores the audio functionality that was broken since Linux version 6.6.y by adding the missing probe functions for the player and reader components. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Specifically, the probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. My patch correctly reinserts the missing probe entries, restoring the intended functionality. The patch modifies the following files: - `sound/soc/sti/sti_uniperif.c`: Changes the visibility of `sti_uniperiph_dai_probe` to non-static. - `sound/soc/sti/uniperif.h`: Adds the declaration of `sti_uniperiph_dai_probe`. - `sound/soc/sti/uniperif_player.c`: Adds `probe` function to `uni_player_dai_ops`. - `sound/soc/sti/uniperif_reader.c`: Adds `probe` function to `uni_reader_dai_ops`. This ensures the correct `probe` functions are utilized, thus fixing the audio regression. --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

2 1

[PATCH v2] ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
by wangdicheng 22 Jul '24

22 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> Sonix HD USB Camera does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. (snip) [1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd [1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00 [1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [1.984127] usb 3-3: Product: USB 2.0 Camera [1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd. [5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v1 -> v2: Arrange the ID in order --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 58156fbca02c..54b2d2f314ac 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2225,6 +2225,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

2 1

aplay: Print '=== PAUSE ===' only if it is supported
by GitHub pull_request - opened 22 Jul '24

22 Jul '24

alsa-project/alsa-utils pull request #271 was opened from ujfalusi: Instead of printing the '=== PAUSE ===' unconditionally before calling do_pause(), move it to the function and only print it if the stream can be paused. If the stream cannot be paused that the '=== PAUSE ===' will be replaced by `PAUSE command ignored (no hw support)` immediately, which is not observable by users but automation scripts will catch the '=== PAUSE ===' and might think that the stream is indeed got paused. Move the print into do_pause() function after the snd_pcm_pause() have returned without error to make sure it is only printed if the stream is paused and we are waiting for the pause release from user to proceed. Request URL : https://github.com/alsa-project/alsa-utils/pull/271 Patch URL : https://github.com/alsa-project/alsa-utils/pull/271.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Fwd: [PATCH] ALSA: hda/realtek: Implement sound init sequence for Samsung Galaxy Book3 Pro 360
by Nick Weihs 22 Jul '24

22 Jul '24

Samsung Galaxy Book3 Pro 360 sends a large amount of data to the codec through hda processing coefficients. This data was captured using a modified version of QEMU, but the actual content of the data remains opaque to me. Elliding any part of the data seems to cause sound to not work. Signed-off-by: Nick Weihs <nick.weihs(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 305 ++++++++++++++++++++++++++++++++++ 1 file changed, 305 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa76d1c88589..cbe6d9689fc3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4796,6 +4796,304 @@ static void alc298_fixup_samsung_amp(struct hda_codec *codec, } } +static inline void alc298_samsung_write_coef_pack2(struct hda_codec *codec, + const unsigned short coefs[4]) +{ + int i; + + for (i = 0; i < 100; i++) { + if ((alc_read_coef_idx(codec, 0x26) & 0x0010) == 0) + break; + } + + alc_write_coef_idx(codec, 0x23, coefs[0]); + alc_write_coef_idx(codec, 0x24, coefs[1]); + alc_write_coef_idx(codec, 0x25, coefs[2]); + alc_write_coef_idx(codec, 0x26, coefs[3]); +} + +static inline void alc298_samsung_write_coef_pack_seq( + struct hda_codec *codec, + unsigned char target, + const unsigned short (*sequence)[4], + int count) +{ + alc_write_coef_idx(codec, 0x22, target); + for (int i = 0; i < count; i++) + alc298_samsung_write_coef_pack2(codec, sequence[i]); +} + +struct alc298_samsung_coeff_fixup_desc { + unsigned char coeff_idx; + unsigned short coeff_value; +}; + +static void alc298_fixup_samsung_amp2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + int i; + static const struct alc298_samsung_coeff_fixup_desc fixups1[] = { + { 0x99, 0x8000 }, { 0x82, 0x4408 }, { 0x32, 0x3f00 }, { 0x0e, 0x6f80 }, + { 0x10, 0x0e21 }, { 0x55, 0x8000 }, { 0x08, 0x2fcf }, { 0x08, 0x2fcf }, + { 0x2d, 0xc020 }, { 0x19, 0x0017 }, { 0x50, 0x1000 }, { 0x0e, 0x6f80 }, + { 0x08, 0x2fcf }, { 0x80, 0x0011 }, { 0x2b, 0x0c10 }, { 0x2d, 0xc020 }, + { 0x03, 0x0042 }, { 0x0f, 0x0062 }, { 0x08, 0x2fcf }, + }; + + static const unsigned short amp_0x38[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0001, 0xB011 }, { 0x201D, 0x0000, 0x0001, 0xB011 }, + { 0x201F, 0x0000, 0x00FE, 0xB011 }, { 0x2021, 0x0000, 0x0000, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x2399, 0x0000, 0x0003, 0xB011 }, + { 0x23A4, 0x0000, 0x00B5, 0xB011 }, { 0x23A5, 0x0000, 0x0001, 0xB011 }, + { 0x23BA, 0x0000, 0x0094, 0xB011 }, { 0x2100, 0x00D0, 0x950E, 0xB017 }, + { 0x2104, 0x0061, 0xD4E2, 0xB017 }, { 0x2108, 0x00D0, 0x950E, 0xB017 }, + { 0x210C, 0x0075, 0xF4E2, 0xB017 }, { 0x2110, 0x00B4, 0x4B0D, 0xB017 }, + { 0x2114, 0x000A, 0x1000, 0xB017 }, { 0x2118, 0x0015, 0x2000, 0xB017 }, + { 0x211C, 0x000A, 0x1000, 0xB017 }, { 0x2120, 0x0075, 0xF4E2, 0xB017 }, + { 0x2124, 0x00B4, 0x4B0D, 0xB017 }, { 0x2128, 0x0000, 0x0010, 0xB017 }, + { 0x212C, 0x0000, 0x0000, 0xB017 }, { 0x2130, 0x0000, 0x0000, 0xB017 }, + { 0x2134, 0x0000, 0x0000, 0xB017 }, { 0x2138, 0x0000, 0x0000, 0xB017 }, + { 0x213C, 0x0000, 0x0010, 0xB017 }, { 0x2140, 0x0000, 0x0000, 0xB017 }, + { 0x2144, 0x0000, 0x0000, 0xB017 }, { 0x2148, 0x0000, 0x0000, 0xB017 }, + { 0x214C, 0x0000, 0x0000, 0xB017 }, { 0x2150, 0x0000, 0x0010, 0xB017 }, + { 0x2154, 0x0000, 0x0000, 0xB017 }, { 0x2158, 0x0000, 0x0000, 0xB017 }, + { 0x215C, 0x0000, 0x0000, 0xB017 }, { 0x2160, 0x0000, 0x0000, 0xB017 }, + { 0x2164, 0x0000, 0x0010, 0xB017 }, { 0x2168, 0x0000, 0x0000, 0xB017 }, + { 0x216C, 0x0000, 0x0000, 0xB017 }, { 0x2170, 0x0000, 0x0000, 0xB017 }, + { 0x2174, 0x0000, 0x0000, 0xB017 }, { 0x2178, 0x0000, 0x0010, 0xB017 }, + { 0x217C, 0x0000, 0x0000, 0xB017 }, { 0x2180, 0x0000, 0x0000, 0xB017 }, + { 0x2184, 0x0000, 0x0000, 0xB017 }, { 0x2188, 0x0000, 0x0000, 0xB017 }, + { 0x218C, 0x0064, 0x5800, 0xB017 }, { 0x2190, 0x00C8, 0xB000, 0xB017 }, + { 0x2194, 0x0064, 0x5800, 0xB017 }, { 0x2198, 0x003D, 0x5BE7, 0xB017 }, + { 0x219C, 0x0054, 0x060A, 0xB017 }, { 0x21A0, 0x00C8, 0xA310, 0xB017 }, + { 0x21A4, 0x0029, 0x4DE5, 0xB017 }, { 0x21A8, 0x0032, 0x420C, 0xB017 }, + { 0x21AC, 0x0029, 0x4DE5, 0xB017 }, { 0x21B0, 0x00FA, 0xE50C, 0xB017 }, + { 0x21B4, 0x0000, 0x0010, 0xB017 }, { 0x21B8, 0x0000, 0x0000, 0xB017 }, + { 0x21BC, 0x0000, 0x0000, 0xB017 }, { 0x21C0, 0x0000, 0x0000, 0xB017 }, + { 0x21C4, 0x0000, 0x0000, 0xB017 }, { 0x21C8, 0x0056, 0xC50F, 0xB017 }, + { 0x21CC, 0x007B, 0xD7E1, 0xB017 }, { 0x21D0, 0x0077, 0xA70E, 0xB017 }, + { 0x21D4, 0x00E0, 0xBDE1, 0xB017 }, { 0x21D8, 0x0032, 0x530E, 0xB017 }, + { 0x2204, 0x00FB, 0x7E0F, 0xB017 }, { 0x2208, 0x000B, 0x02E1, 0xB017 }, + { 0x220C, 0x00FB, 0x7E0F, 0xB017 }, { 0x2210, 0x00D5, 0x17E1, 0xB017 }, + { 0x2214, 0x00C0, 0x130F, 0xB017 }, { 0x2218, 0x00E5, 0x0A00, 0xB017 }, + { 0x221C, 0x00CB, 0x1500, 0xB017 }, { 0x2220, 0x00E5, 0x0A00, 0xB017 }, + { 0x2224, 0x00D5, 0x17E1, 0xB017 }, { 0x2228, 0x00C0, 0x130F, 0xB017 }, + { 0x222C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2230, 0x0017, 0x48E2, 0xB017 }, + { 0x2234, 0x00F5, 0xDB0E, 0xB017 }, { 0x2238, 0x00EF, 0x5CE2, 0xB017 }, + { 0x223C, 0x00C1, 0xCC0D, 0xB017 }, { 0x2240, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2244, 0x0017, 0x48E2, 0xB017 }, { 0x2248, 0x00F5, 0xDB0E, 0xB017 }, + { 0x224C, 0x00EF, 0x5CE2, 0xB017 }, { 0x2250, 0x00C1, 0xCC0D, 0xB017 }, + { 0x2254, 0x00F5, 0xDB0E, 0xB017 }, { 0x2258, 0x0017, 0x48E2, 0xB017 }, + { 0x225C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2260, 0x00EF, 0x5CE2, 0xB017 }, + { 0x2264, 0x00C1, 0xCC0D, 0xB017 }, { 0x2268, 0x00F5, 0xDB0E, 0xB017 }, + { 0x226C, 0x0017, 0x48E2, 0xB017 }, { 0x2270, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2274, 0x00EF, 0x5CE2, 0xB017 }, { 0x2278, 0x00C1, 0xCC0D, 0xB017 }, + { 0x227C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2280, 0x0017, 0x48E2, 0xB017 }, + { 0x2284, 0x00F5, 0xDB0E, 0xB017 }, { 0x2288, 0x00EF, 0x5CE2, 0xB017 }, + { 0x228C, 0x00C1, 0xCC0D, 0xB017 }, { 0x22CC, 0x00E8, 0x8D00, 0xB017 }, + { 0x22D0, 0x0000, 0x0000, 0xB017 }, { 0x22D4, 0x0018, 0x72FF, 0xB017 }, + { 0x22D8, 0x00CE, 0x25E1, 0xB017 }, { 0x22DC, 0x002F, 0xE40E, 0xB017 }, + { 0x238E, 0x0000, 0x0099, 0xB011 }, { 0x238F, 0x0000, 0x0011, 0xB011 }, + { 0x2390, 0x0000, 0x0056, 0xB011 }, { 0x2391, 0x0000, 0x0004, 0xB011 }, + { 0x2392, 0x0000, 0x00BB, 0xB011 }, { 0x2393, 0x0000, 0x006D, 0xB011 }, + { 0x2394, 0x0000, 0x0010, 0xB011 }, { 0x2395, 0x0000, 0x0064, 0xB011 }, + { 0x2396, 0x0000, 0x00B6, 0xB011 }, { 0x2397, 0x0000, 0x0028, 0xB011 }, + { 0x2398, 0x0000, 0x000B, 0xB011 }, { 0x239A, 0x0000, 0x0099, 0xB011 }, + { 0x239B, 0x0000, 0x000D, 0xB011 }, { 0x23A6, 0x0000, 0x0064, 0xB011 }, + { 0x23A7, 0x0000, 0x0078, 0xB011 }, { 0x23B9, 0x0000, 0x0000, 0xB011 }, + { 0x23E0, 0x0000, 0x0021, 0xB011 }, { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x39[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0002, 0xB011 }, { 0x201D, 0x0000, 0x0002, 0xB011 }, + { 0x201F, 0x0000, 0x00FD, 0xB011 }, { 0x2021, 0x0000, 0x0001, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x2399, 0x0000, 0x0003, 0xB011 }, + { 0x23A4, 0x0000, 0x00B5, 0xB011 }, { 0x23A5, 0x0000, 0x0001, 0xB011 }, + { 0x23BA, 0x0000, 0x0094, 0xB011 }, { 0x2100, 0x00D0, 0x950E, 0xB017 }, + { 0x2104, 0x0061, 0xD4E2, 0xB017 }, { 0x2108, 0x00D0, 0x950E, 0xB017 }, + { 0x210C, 0x0075, 0xF4E2, 0xB017 }, { 0x2110, 0x00B4, 0x4B0D, 0xB017 }, + { 0x2114, 0x000A, 0x1000, 0xB017 }, { 0x2118, 0x0015, 0x2000, 0xB017 }, + { 0x211C, 0x000A, 0x1000, 0xB017 }, { 0x2120, 0x0075, 0xF4E2, 0xB017 }, + { 0x2124, 0x00B4, 0x4B0D, 0xB017 }, { 0x2128, 0x0000, 0x0010, 0xB017 }, + { 0x212C, 0x0000, 0x0000, 0xB017 }, { 0x2130, 0x0000, 0x0000, 0xB017 }, + { 0x2134, 0x0000, 0x0000, 0xB017 }, { 0x2138, 0x0000, 0x0000, 0xB017 }, + { 0x213C, 0x0000, 0x0010, 0xB017 }, { 0x2140, 0x0000, 0x0000, 0xB017 }, + { 0x2144, 0x0000, 0x0000, 0xB017 }, { 0x2148, 0x0000, 0x0000, 0xB017 }, + { 0x214C, 0x0000, 0x0000, 0xB017 }, { 0x2150, 0x0000, 0x0010, 0xB017 }, + { 0x2154, 0x0000, 0x0000, 0xB017 }, { 0x2158, 0x0000, 0x0000, 0xB017 }, + { 0x215C, 0x0000, 0x0000, 0xB017 }, { 0x2160, 0x0000, 0x0000, 0xB017 }, + { 0x2164, 0x0000, 0x0010, 0xB017 }, { 0x2168, 0x0000, 0x0000, 0xB017 }, + { 0x216C, 0x0000, 0x0000, 0xB017 }, { 0x2170, 0x0000, 0x0000, 0xB017 }, + { 0x2174, 0x0000, 0x0000, 0xB017 }, { 0x2178, 0x0000, 0x0010, 0xB017 }, + { 0x217C, 0x0000, 0x0000, 0xB017 }, { 0x2180, 0x0000, 0x0000, 0xB017 }, + { 0x2184, 0x0000, 0x0000, 0xB017 }, { 0x2188, 0x0000, 0x0000, 0xB017 }, + { 0x218C, 0x0064, 0x5800, 0xB017 }, { 0x2190, 0x00C8, 0xB000, 0xB017 }, + { 0x2194, 0x0064, 0x5800, 0xB017 }, { 0x2198, 0x003D, 0x5BE7, 0xB017 }, + { 0x219C, 0x0054, 0x060A, 0xB017 }, { 0x21A0, 0x00C8, 0xA310, 0xB017 }, + { 0x21A4, 0x0029, 0x4DE5, 0xB017 }, { 0x21A8, 0x0032, 0x420C, 0xB017 }, + { 0x21AC, 0x0029, 0x4DE5, 0xB017 }, { 0x21B0, 0x00FA, 0xE50C, 0xB017 }, + { 0x21B4, 0x0000, 0x0010, 0xB017 }, { 0x21B8, 0x0000, 0x0000, 0xB017 }, + { 0x21BC, 0x0000, 0x0000, 0xB017 }, { 0x21C0, 0x0000, 0x0000, 0xB017 }, + { 0x21C4, 0x0000, 0x0000, 0xB017 }, { 0x21C8, 0x0056, 0xC50F, 0xB017 }, + { 0x21CC, 0x007B, 0xD7E1, 0xB017 }, { 0x21D0, 0x0077, 0xA70E, 0xB017 }, + { 0x21D4, 0x00E0, 0xBDE1, 0xB017 }, { 0x21D8, 0x0032, 0x530E, 0xB017 }, + { 0x2204, 0x00FB, 0x7E0F, 0xB017 }, { 0x2208, 0x000B, 0x02E1, 0xB017 }, + { 0x220C, 0x00FB, 0x7E0F, 0xB017 }, { 0x2210, 0x00D5, 0x17E1, 0xB017 }, + { 0x2214, 0x00C0, 0x130F, 0xB017 }, { 0x2218, 0x00E5, 0x0A00, 0xB017 }, + { 0x221C, 0x00CB, 0x1500, 0xB017 }, { 0x2220, 0x00E5, 0x0A00, 0xB017 }, + { 0x2224, 0x00D5, 0x17E1, 0xB017 }, { 0x2228, 0x00C0, 0x130F, 0xB017 }, + { 0x222C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2230, 0x0017, 0x48E2, 0xB017 }, + { 0x2234, 0x00F5, 0xDB0E, 0xB017 }, { 0x2238, 0x00EF, 0x5CE2, 0xB017 }, + { 0x223C, 0x00C1, 0xCC0D, 0xB017 }, { 0x2240, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2244, 0x0017, 0x48E2, 0xB017 }, { 0x2248, 0x00F5, 0xDB0E, 0xB017 }, + { 0x224C, 0x00EF, 0x5CE2, 0xB017 }, { 0x2250, 0x00C1, 0xCC0D, 0xB017 }, + { 0x2254, 0x00F5, 0xDB0E, 0xB017 }, { 0x2258, 0x0017, 0x48E2, 0xB017 }, + { 0x225C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2260, 0x00EF, 0x5CE2, 0xB017 }, + { 0x2264, 0x00C1, 0xCC0D, 0xB017 }, { 0x2268, 0x00F5, 0xDB0E, 0xB017 }, + { 0x226C, 0x0017, 0x48E2, 0xB017 }, { 0x2270, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2274, 0x00EF, 0x5CE2, 0xB017 }, { 0x2278, 0x00C1, 0xCC0D, 0xB017 }, + { 0x227C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2280, 0x0017, 0x48E2, 0xB017 }, + { 0x2284, 0x00F5, 0xDB0E, 0xB017 }, { 0x2288, 0x00EF, 0x5CE2, 0xB017 }, + { 0x228C, 0x00C1, 0xCC0D, 0xB017 }, { 0x22CC, 0x00E8, 0x8D00, 0xB017 }, + { 0x22D0, 0x0000, 0x0000, 0xB017 }, { 0x22D4, 0x0018, 0x72FF, 0xB017 }, + { 0x22D8, 0x00CE, 0x25E1, 0xB017 }, { 0x22DC, 0x002F, 0xE40E, 0xB017 }, + { 0x238E, 0x0000, 0x0099, 0xB011 }, { 0x238F, 0x0000, 0x0011, 0xB011 }, + { 0x2390, 0x0000, 0x0056, 0xB011 }, { 0x2391, 0x0000, 0x0004, 0xB011 }, + { 0x2392, 0x0000, 0x00BB, 0xB011 }, { 0x2393, 0x0000, 0x006D, 0xB011 }, + { 0x2394, 0x0000, 0x0010, 0xB011 }, { 0x2395, 0x0000, 0x0064, 0xB011 }, + { 0x2396, 0x0000, 0x00B6, 0xB011 }, { 0x2397, 0x0000, 0x0028, 0xB011 }, + { 0x2398, 0x0000, 0x000B, 0xB011 }, { 0x239A, 0x0000, 0x0099, 0xB011 }, + { 0x239B, 0x0000, 0x000D, 0xB011 }, { 0x23A6, 0x0000, 0x0064, 0xB011 }, + { 0x23A7, 0x0000, 0x0078, 0xB011 }, { 0x23B9, 0x0000, 0x0000, 0xB011 }, + { 0x23E0, 0x0000, 0x0021, 0xB011 }, { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x3c[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0001, 0xB011 }, { 0x201D, 0x0000, 0x0001, 0xB011 }, + { 0x201F, 0x0000, 0x00FE, 0xB011 }, { 0x2021, 0x0000, 0x0000, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x23BA, 0x0000, 0x008D, 0xB011 }, + { 0x2128, 0x0005, 0x460D, 0xB017 }, { 0x212C, 0x00F6, 0x73E5, 0xB017 }, + { 0x2130, 0x0005, 0x460D, 0xB017 }, { 0x2134, 0x00C0, 0xE9E5, 0xB017 }, + { 0x2138, 0x00D5, 0x010B, 0xB017 }, { 0x213C, 0x009D, 0x7809, 0xB017 }, + { 0x2140, 0x00C5, 0x0EED, 0xB017 }, { 0x2144, 0x009D, 0x7809, 0xB017 }, + { 0x2148, 0x00C4, 0x4EF0, 0xB017 }, { 0x214C, 0x003A, 0x3106, 0xB017 }, + { 0x2150, 0x00AF, 0x750E, 0xB017 }, { 0x2154, 0x008C, 0x1FF1, 0xB017 }, + { 0x2158, 0x009E, 0x360C, 0xB017 }, { 0x215C, 0x008C, 0x1FF1, 0xB017 }, + { 0x2160, 0x004D, 0xAC0A, 0xB017 }, { 0x2164, 0x007D, 0xA00F, 0xB017 }, + { 0x2168, 0x00E1, 0x9CE3, 0xB017 }, { 0x216C, 0x00E8, 0x590E, 0xB017 }, + { 0x2170, 0x00E1, 0x9CE3, 0xB017 }, { 0x2174, 0x0066, 0xFA0D, 0xB017 }, + { 0x2178, 0x0000, 0x0010, 0xB017 }, { 0x217C, 0x0000, 0x0000, 0xB017 }, + { 0x2180, 0x0000, 0x0000, 0xB017 }, { 0x2184, 0x0000, 0x0000, 0xB017 }, + { 0x2188, 0x0000, 0x0000, 0xB017 }, { 0x218C, 0x0000, 0x0010, 0xB017 }, + { 0x2190, 0x0000, 0x0000, 0xB017 }, { 0x2194, 0x0000, 0x0000, 0xB017 }, + { 0x2198, 0x0000, 0x0000, 0xB017 }, { 0x219C, 0x0000, 0x0000, 0xB017 }, + { 0x21A0, 0x0000, 0x0010, 0xB017 }, { 0x21A4, 0x0000, 0x0000, 0xB017 }, + { 0x21A8, 0x0000, 0x0000, 0xB017 }, { 0x21AC, 0x0000, 0x0000, 0xB017 }, + { 0x21B0, 0x0000, 0x0000, 0xB017 }, { 0x21B4, 0x0000, 0x0010, 0xB017 }, + { 0x21B8, 0x0000, 0x0000, 0xB017 }, { 0x21BC, 0x0000, 0x0000, 0xB017 }, + { 0x21C0, 0x0000, 0x0000, 0xB017 }, { 0x21C4, 0x0000, 0x0000, 0xB017 }, + { 0x23B9, 0x0000, 0x0000, 0xB011 }, { 0x23E0, 0x0000, 0x0020, 0xB011 }, + { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x3d[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0002, 0xB011 }, { 0x201D, 0x0000, 0x0002, 0xB011 }, + { 0x201F, 0x0000, 0x00FD, 0xB011 }, { 0x2021, 0x0000, 0x0001, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x23BA, 0x0000, 0x008D, 0xB011 }, + { 0x2128, 0x0005, 0x460D, 0xB017 }, { 0x212C, 0x00F6, 0x73E5, 0xB017 }, + { 0x2130, 0x0005, 0x460D, 0xB017 }, { 0x2134, 0x00C0, 0xE9E5, 0xB017 }, + { 0x2138, 0x00D5, 0x010B, 0xB017 }, { 0x213C, 0x009D, 0x7809, 0xB017 }, + { 0x2140, 0x00C5, 0x0EED, 0xB017 }, { 0x2144, 0x009D, 0x7809, 0xB017 }, + { 0x2148, 0x00C4, 0x4EF0, 0xB017 }, { 0x214C, 0x003A, 0x3106, 0xB017 }, + { 0x2150, 0x00AF, 0x750E, 0xB017 }, { 0x2154, 0x008C, 0x1FF1, 0xB017 }, + { 0x2158, 0x009E, 0x360C, 0xB017 }, { 0x215C, 0x008C, 0x1FF1, 0xB017 }, + { 0x2160, 0x004D, 0xAC0A, 0xB017 }, { 0x2164, 0x007D, 0xA00F, 0xB017 }, + { 0x2168, 0x00E1, 0x9CE3, 0xB017 }, { 0x216C, 0x00E8, 0x590E, 0xB017 }, + { 0x2170, 0x00E1, 0x9CE3, 0xB017 }, { 0x2174, 0x0066, 0xFA0D, 0xB017 }, + { 0x2178, 0x0000, 0x0010, 0xB017 }, { 0x217C, 0x0000, 0x0000, 0xB017 }, + { 0x2180, 0x0000, 0x0000, 0xB017 }, { 0x2184, 0x0000, 0x0000, 0xB017 }, + { 0x2188, 0x0000, 0x0000, 0xB017 }, { 0x218C, 0x0000, 0x0010, 0xB017 }, + { 0x2190, 0x0000, 0x0000, 0xB017 }, { 0x2194, 0x0000, 0x0000, 0xB017 }, + { 0x2198, 0x0000, 0x0000, 0xB017 }, { 0x219C, 0x0000, 0x0000, 0xB017 }, + { 0x21A0, 0x0000, 0x0010, 0xB017 }, { 0x21A4, 0x0000, 0x0000, 0xB017 }, + { 0x21A8, 0x0000, 0x0000, 0xB017 }, { 0x21AC, 0x0000, 0x0000, 0xB017 }, + { 0x21B0, 0x0000, 0x0000, 0xB017 }, { 0x21B4, 0x0000, 0x0010, 0xB017 }, + { 0x21B8, 0x0000, 0x0000, 0xB017 }, { 0x21BC, 0x0000, 0x0000, 0xB017 }, + { 0x21C0, 0x0000, 0x0000, 0xB017 }, { 0x21C4, 0x0000, 0x0000, 0xB017 }, + { 0x23B9, 0x0000, 0x0000, 0xB011 }, { 0x23E0, 0x0000, 0x0020, 0xB011 }, + { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_seq1[][4] = { + { 0x23FF, 0x0000, 0x0000, 0xB011 }, { 0x203A, 0x0000, 0x0080, 0xB011 }, + }; + + static const struct alc298_samsung_coeff_fixup_desc fixups2[] = { + { 0x4f, 0xb029 }, { 0x05, 0x2be0 }, { 0x30, 0x2421 }, + }; + + + static const unsigned short amp_seq2[][4] = { + { 0x203A, 0x0000, 0x0081, 0xB011 }, { 0x23FF, 0x0000, 0x0001, 0xB011 }, + }; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + ///// First set of fixups + for (i = 0; i < ARRAY_SIZE(fixups1); i++) + alc_write_coef_idx(codec, fixups1[i].coeff_idx, fixups1[i].coeff_value); + + ///// First set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_0x38, ARRAY_SIZE(amp_0x38)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_0x39, ARRAY_SIZE(amp_0x39)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_0x3c, ARRAY_SIZE(amp_0x3c)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_0x3d, ARRAY_SIZE(amp_0x3d)); + + ///// Second set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq1, ARRAY_SIZE(amp_seq1)); + + ////// Second set of fixups + for (i = 0; i < ARRAY_SIZE(fixups2); i++) + alc_write_coef_idx(codec, fixups2[i].coeff_idx, fixups2[i].coeff_value); + + ///// Third set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq2, ARRAY_SIZE(amp_seq2)); + + ///// Final fixup + alc_write_coef_idx(codec, 0x10, 0x0F21); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -7426,6 +7724,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC298_FIXUP_SAMSUNG_AMP, + ALC298_FIXUP_SAMSUNG_AMP2, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -9021,6 +9320,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET }, + [ALC298_FIXUP_SAMSUNG_AMP2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_samsung_amp2 + }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10352,6 +10655,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG-KB1US)", ALC298_FIXUP_SAMSUNG_AMP2), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -10781,6 +11085,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_AMP, .name = "alc298-samsung-amp"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP2, .name = "alc298-samsung-amp2"}, {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, -- 2.45.2

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[PATCH v10] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by Baojun Xu 21 Jul '24

21 Jul '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. The code support Realtek as the primary codec. In patch version-10, add multi devices firmware binary support, to compatble with windows driver, they can share same firmware binary. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v10: - Remove project id 103c8b93 in Patch_realtek.c, as it was used for test only, offcial project id is 103c8daa (Gemtree). - Add TAS2781_REG_CLK_CONFIG for check if AMP has been reset. - Add table for pre-reset and post-reset. - Perform pre-reset config before reset AMP. - Perform post-reset config after reset AMP. - Fix wrong offset in calibration data issue. - Remove tuning switch after firmware download. - Force stop play in runtime suspend. - Apply patch for component change in bind & unbind. - Remove program download and calibration data apply, switch tuning only in runtime resume. - Check if AMP has reseted in system resume, and do firmware download and calibration data apply. - Change flag 0xCx to 0x8x for coeff data force download. - Add device index for multi device firmware binary file compatible. - Check device index before registers write during firmware download. - Force config data download after program download. - Remove config download in tuning switch. v9: - Add parentheses for variables in micro define. - Add define for calibration registers address identify. - Change variable name from calibration_data to cali_data. - Add variable for calibration registers address in struct tasdevice_priv. - Change format for alignment to parentheses. - Remove tas2781_reg_defaults as it will be over write in firmware download. - Remove RESET for second device in tas2781_spi_reset(), as it will cause no sound on second device. - Add calibration registers address initial in tasdevice_spi_init(). - Remove unnecessary parentheses. - Remove exception in tas2781_hda_playback_hook(). - Remove hard code for calibration registers address. - Add support for calibration data format V3 for registers address identify, as new firmware tool will define calibration registers address dynamic, so can't hard code it, need read it from BIOS with calibration data. - Remove some debug print. - Change hard code value (3) to ARRAY_SIZE(tas2781_snd_controls)/2. - New patch apply for change position of component_del(). - Change label from out, out1 to config_err and block_err. - Remove unnecessary parentheses in tas2781_spi_fwlib.c. - Remove tasdevice_spi_prmg_calibdata_load(). v8: - Move *-y position in sound/pci/hda/Makefile. - Change + to | for bit operation in macro define. - Change TASDEVICE_Checksum to TASDEVICE_CHECKSUM. v7: - Add new project Gemtree (0x103c8daa) support for tas2781 spi driver. v6: - Remove client in tasdevice_priv struct as it was not used. - Change format for tasdevice_spi_dev_write() in tas2781_hda_spi.c. - Change update bits function to read & write mode, as tas2781 use last bit to mask read & write, it cause regmap_update_bits work not as our expected. - Add element in tas2781_snd_controls, tas2781_prof_ctrl, tas2781_dsp_conf_ctrl, to support second device. Previous version add sound card for first index only. - Change calibration registers address declaire to TASDEVICE_REG() mode. - Add snd_ctl_new1() for second device. - Change format for tas2781_hda_unbind() to fit 79 chars in one line. - Remove client for spi as it was not used. - Remove unused variables. - Remove cur_prog clean in runtime_suspend(). - Remove unused variables. - Add tasdevice_spi_tuning_switch() in runtime_resume. - Remove cur_prog clean in system_resume(). - Add tuning switch if playback started in system_resume(). - Change TASDEVICE_PAGE_REG to TASDEVICE_REG_ID in tas2781_spi_fwlib.c. - Remove calibration binary load functions as data will be loaded from EFI. v5: - Combined all modification into one patch for pass the compile. - Remove old hardware id "TIAS2781" as no production on market. - Change max page size to 256 in tas2781-spi.h. - Change book reg from 127 to GENMASK(7, 1), as one bit shift is needed. - Change register address define as one bit shift needed for offset. - Change block check sum comments to Firmware block from I2C. - Change define of (page + reg) to TASDEVICE_PAGE_REG from TASDEVICE_PGRG. - Change to lower case in defined value. - Add registers default value table in tas2781_hda_spi.c - Change rang_max to 256 * 256. - Add zero_flag_mask in regmap_config. - Change max_register to 256 * 256. - Add registers default table into structure regmap_config. - Change parameter from book number to whole register address. - Remove page 0 switch as SPI driver will do it. - Add bit(0) for read function as tas2781 will check last bit of address, to determine operation mode is read (1) or write (0). - Change switch case to if check in tas2781_hda_playback_hook(), as it may cause compile problem in clang-19 building. - Change variable declaration position. - Change order in if check. - Remove old hardware id ("TIAS2781") support. - Remove debug message in runtime suspend and resume. - Remove regmap.h include in tas2781_spi_fwlib.c. - Remove MODULE_ in fwlib as this file will work with tas2781_hda_spi.c only. v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs - Move include of sound/tas2781-dsp.h to c source file tas2781_hda_spi.c, tas2781_spi_fwlib.c, as it's not needed in header file tas2781-spi.h - Use page and book number define in other micro define - Change 4000000 to (4 * HZ_PER_MHZ) - Add define for calibration variable name and size in UEFI. - Remove structure tasdevice & tasdevice_irqinfo, and move variables in to structure tasdevice_priv - Remove some unused variables in sttructure tasdevice_priv - Remove function declare of tascodec_init(), use tascodec_spi_init() - Remove function declare of tasdevice_remove() - Add array_size.h included in tas2781_hda_spi.c - Add cleanup.h included for change mutex to guard - Add units.h included - Include tas2781-dsp.h - Remove unused variables in tas2781_hda structure - Change 0xff to GENMASK(7, 0) - Change 128 to define of max page size - Change code as all variables was moved to tasdevice_priv structure - Change comments for error in book write, or page write - Remove initial of ret, and return 0 directly - Change comments for wrong typo, add space in front and end - Add check for bulk read, max size should be a page size - Change usleep_rang() to fsleep() - Change mutex_lock to guard(mutex), and remove mutex_unlock() - Change tasdevice_spi_init() to void from int as no value return required - Change second parameter to sizeof - Remove tasdevice_clamp(), use clamp() - Add compatible with old hardware id "TIAS2781" - Remove cs_gpio and spi device in tas2781_read_acpi() as use default CS - Change dev-index check, return -EINVAL if it's <= 0 - Change 0xFF to U8_MAX - Remove GPIO get for chip-select, use default from SPI device - Perform RESET in first AMP only as all AMPs share same RESET pin - return 0 directly in some simple routine to avoid variable initial - Change comments for function which was used by system - Change "ON" "OFF" to function str_on_off(...) - Change format of GUID - Add temp buffer for first efi.get_variable() - Free data if it was allocated - Change format of debug message of calibration date time print - Remove total_sz = 0, as it's not needed - Move save_calibration to after tuning_switch to avoid calibration data overwrite - Remove dev from structure tasdevice_hda, use dev in tasdevice_priv - Remove 0 from { "tas2781-hda", 0}, & {"TXNW2781", 0}, - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v3: - Move HID up to above /* Non-conforming _HID ... */ in scan.c, for avoid misunderstanding. - Move HID up to above /* Non-conforming _HID ... */ in serial-multi-instantiate.c, for avoid misunderstanding. - Change objs to y for snd-hda-scodec-tas2781-spi- in Makefile. - Change format for some define in tas2781-spi.h. - Add define for TASDEVICE_MAX_BOOK_NUM, TASDEVICE_MAX_PAGE. - Move declare of "enum calib_data {" to header file from source file. - Remove "enum device_catlog_id {" as only one platform was supported now. - Remove "struct calidata", as we will save pure calibration data only. - Remove "struct calidata", "enum device_catlog_id" in tasdevice_priv. - Add calibration_data in tasdevice_priv to save pure calibration data. - Remove declare of tasdevice_save_calibration() and tasdevice_apply_calibration(), as it will be used within same c file. - Add below header files included in tas2781_hda_spi.c: bits.h, mutex.h, property.h, time.h, types.h. - Move two micro define to header file tas2781-spi.h. - Change format of some micro define. - Change format of some structure. - Remove irq in tas2781_hda, as have it already in tasdevice_priv. - Remove some local functions declare as not necessarily. - Return error if regmap_write() failed. - Change fixed value 2 to sizeof(data). - Remove all of EXPORT_SYMBOL_GPL, as all of function was called within same module. - Remove empty line after last return in some functions. - Remove some variable initialyzation. - Remove variable sub, store acpi_subsystem_id directly. - Remove wrong comments for device index, it's must, can't be NULL. - Remove some local variables, use elements in structure directly. - Append commas in last element of an array. - Change calibration data process, didn't save all of calibration data in EFI, get in local, and save pure calibration data in tasdevice_priv. - Call tasdevice_save_calibration() from function pointer in tasdevice_priv; - Remove subid as one platform was supported only, needn't check. - Add initialyzation for local variable. - Move regmap initialyzation to before acpi read. - Call tasdevice_apply_calibration() from function pointer in tasdevice_priv; - Remove MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB), as all functions was performed within same module. - Update format and variables declare order. - Change format of multi conditions for if. - Remove casting which is not needed. - Change variables type to avoid casting. - Remove Unneeded parentheses. - Change if check to avoid goto. - Add {} for keep same style. - Remove some local variables, use elements in structure directly. v2: - Change depends on SPI to SPI_MASTER in sound/pci/hda/Kconfig - Add select SND_HDA_SCODEC_COMPONENT in sound/pci/hda/Kconfig - Change comp_generic_fixup() 5th parameter from "-%s:00" to "-%s:00-tas2781-hda.%d" in sound/pci/hda/patch_realtek.c - Change chain_id from ALC285_FIXUP_THINKPAD_HEADSET_JACK to ALC285_FIXUP_HP_GPIO_LED in sound/pci/hda/patch_realtek.c - Remove project name "Gemtree" - Update information in file header of tas2781-spi.h. - Remove useless micro define. - Change TASDEVICE_I2CChecksum to TASDEVICE_Checksum - Remove enum control_bus as current code support SPI only. - Remove device define as current code support TAS2781 only. - Remove spi_device **spi_devs in structure tasdevice_priv. - Remove cal_binaryname in structure tasdevice_priv. - Remove ndev in structure tasdevice_priv. - Change isacpi and isspi, replace by control_bus(I2C or SPI). - Remove void tasdevice_spi_dsp_remove(void *context). - Add acpi_device and irq in structure tas2781_hda in tas2781_hda_spi.c. - Remove parameter chn in all of registers access functions. - Add tascodec_spi_init(). - Add chip select gpio set for SPI device. - Change tasdevice_tuning_switch() to tasdevice_spi_tuning_switch(). - Change device offset from tas_priv->ndev to tas_priv->index. - Change tasdevice_dsp_remove() to tasdevice_spi_dsp_remove(). - tasdevice_prmg_load() to tasdevice_spi_prmg_load(). - Change tasdevice_config_info_remove() to tasdevice_spi_config_info_remove(). - Add one dummy byte shift when read from page 2~127, or not book 0. - Change tasdevice_spi_change_chn_book() to tasdevice_spi_switch_book() - Perform put_device(physdev) before error return. - Change calibration data id to 2781 from 2783 in EFI - Change calibration data structure name in comments. - Remove file name in tas2781_spi_fwlib.c. - Remove #include <linux/i2c.h> as it not needed in SPI mode. - Change TAB to speace in micro define in tas2781_spi_fwlib.c. - Change to up-case for DSP. - Change all of multi-line comments format, empty first line. - Change all of sizeof(struct xx) to sizeof(*xx). - Remove Explicit casting for variables compare to avoid possible fault. - Return directly without goto. - Change 1 << xx to BIT(xx). - Remove deviceNumber[] as current SPI device will support one device only. - Add memory free before return error. - Change "buf[offset]; offset += 1" to buf[offset++]. - Remove some debug information print. - Change firmware binary to single device mode. - Change all bexx_to_cpup() to get_unaligned_bexx(). - Remove ndev process as current SPI device support single device only. - Remove chn and chnend for single device support only. - Change all of registers read/write function, remove parameter chn. - Create new function for single write, burst write, delay, field write in tasdevice_process_block(). --- drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 158 ++ sound/pci/hda/tas2781_hda_spi.c | 1317 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2115 +++++++++++++++++ 8 files changed, 3632 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 59771412686b..b1a8bf1043da 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1769,6 +1769,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 3be016cfe601..b15c819023d0 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -368,6 +368,17 @@ static const struct smi_node cs35l57_hda = { .bus_type = SMI_AUTO_DETECT, }; +static const struct smi_node tas2781_hda = { + .instances = { + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + {} + }, + .bus_type = SMI_AUTO_DETECT, +}; + /* * Note new device-ids must also be added to ignore_serial_bus_ids in * drivers/acpi/scan.c: acpi_device_enumeration_by_parent(). @@ -380,6 +391,7 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "CSC3556", (unsigned long)&cs35l56_hda }, { "CSC3557", (unsigned long)&cs35l57_hda }, { "INT3515", (unsigned long)&int3515_data }, + { "TXNW2781", (unsigned long)&tas2781_hda }, /* Non-conforming _HID for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, { "CLSA0101", (unsigned long)&cs35l41_hda }, diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250..62cd9ba408e2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -206,6 +206,20 @@ config SND_HDA_SCODEC_TAS2781_I2C comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m +config SND_HDA_SCODEC_TAS2781_SPI + tristate "Build TAS2781 HD-audio side codec support for SPI Bus" + depends on SPI_MASTER + depends on ACPI + depends on EFI + depends on SND_SOC + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 SPI HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_SPI=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 80210f845df2..210c406dfbc5 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -40,6 +40,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -72,6 +73,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_SPI) += snd-hda-scodec-tas2781-spi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c3a86a99f8c6..51707270943a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,6 +6924,11 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } +static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7501,6 +7506,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, @@ -9726,6 +9732,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC245_FIXUP_TAS2781_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_spi, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -10263,6 +10275,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8daa, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h new file mode 100644 index 000000000000..0cadfd18bde2 --- /dev/null +++ b/sound/pci/hda/tas2781-spi.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for TAS2781 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2781_SPI_H__ +#define __TAS2781_SPI_H__ + +#define TASDEVICE_RATES \ + (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TASDEVICE_MAX_BOOK_NUM 256 +#define TASDEVICE_MAX_PAGE 256 + +#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) + +/* PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) +#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) +#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) +#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) +#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) +#define TASDEVICE_REG(book, page, reg) \ + (((book) << 16) | ((page) << 8) | ((reg) << 1)) + +/* Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET (0x19) + +/* Block Checksum */ +#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) + +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEVICE_CALIBRATION_DATA_SIZE 256 + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +struct tasdevice_priv { + struct tasdevice_fw *cali_data_fmw; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + const char *acpi_subsystem_id; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + unsigned int err_code; + bool is_loading; + bool is_loaderr; + int cur_book; + int cur_prog; + int cur_conf; + int fw_state; + int index; + int irq; + void *codec; + bool force_fwload_status; + bool playback_started; + unsigned int cali_data[CALIB_MAX]; + unsigned int cali_reg_array[CALIB_MAX]; + + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, + int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, + int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); +}; + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int *value); +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int value); +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, + unsigned int reg, unsigned int mask, + unsigned int value); + +void tasdevice_spi_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_spi_config_info_remove(void *context); +int tasdevice_spi_dsp_parser(void *context); +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_spi_dsp_remove(void *context); +void tasdevice_spi_calbin_remove(void *context); +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, + int rca_conf_no); +int tasdevice_spi_prmg_load(void *context, int prm_no); +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_spi_tuning_switch(void *context, int state); +int tas2781_spi_load_calibration(void *context, char *file_name, + unsigned short i); +#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c new file mode 100644 index 000000000000..dd979cb46fce --- /dev/null +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -0,0 +1,1317 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <linux/acpi.h> +#include <linux/array_size.h> +#include <linux/bits.h> +#include <linux/cleanup.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/pm_runtime.h> +#include <linux/property.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/time.h> +#include <linux/types.h> +#include <linux/units.h> + +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2781-spi.h" + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +struct tas2781_hda { + struct tasdevice_priv *priv; + struct acpi_device *dacpi; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *prof_ctl; +}; + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = TASDEVICE_MAX_SIZE, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = GENMASK(7, 0), + .selector_shift = 0, + .window_start = 0, + .window_len = TASDEVICE_MAX_PAGE, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .zero_flag_mask = true, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = TASDEVICE_MAX_SIZE, +}; + +struct tasdevice_table { + int reg; + unsigned char data[8]; +}; + +static struct tasdevice_table tas2781_pre_reset_table[] = { + /* Values setting for pre-reset. */ + { TASDEVICE_REG(0x00, 0x00, 0x02), {0x01, 0x1a, },}, + { TASDEVICE_REG(0x00, 0x01, 0x37), {0x01, 0x3a, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xc1, },}, + { TASDEVICE_REG(0x00, 0x01, 0x19), {0x01, 0xe0, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xd5, },}, + { TASDEVICE_REG(0x00, 0x00, 0x00), {0x01, 0x00, },}, +}; + +static struct tasdevice_table tas2781_post_reset_table[] = { + /* Values setting for post-reset. */ + { TASDEVICE_REG(0x00, 0x01, 0x37), {0x01, 0x3a, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xc1, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xd5, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0a), {0x01, 0x01, },}, + { TASDEVICE_REG(0x00, 0x00, 0x00), {0x01, 0x00, },}, +}; + +static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) { + dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); + return ret; + } + tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); + } + return ret; +} + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int *val) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char data[2]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, sizeof(data)); + *val = data[1]; + } else { + ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + if (len > TASDEVICE_MAX_PAGE) + len = TASDEVICE_MAX_PAGE; + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[TASDEVICE_MAX_PAGE+1]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, + len + 1); + memcpy(data, buf + 1, len); + } else { + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, + len); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int mask, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret, val; + + /* + * In our TAS2781 SPI mode, read/write was masked in last bit of + * address, it cause regmap_update_bits() not work as expected. + */ + ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) +{ + int ret; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + fsleep(800); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } + for (int j = 0; j >= 0; j++) { + if (tas2781_pre_reset_table[j].reg == 0) + break; + if (tas2781_pre_reset_table[j].data[0] > 1) { + tasdevice_spi_dev_bulk_write(tas_dev, + tas2781_pre_reset_table[j].reg, + &tas2781_pre_reset_table[j].data[1], + tas2781_pre_reset_table[j].data[0]); + } else { + tasdevice_spi_dev_write(tas_dev, + tas2781_pre_reset_table[j].reg, + tas2781_pre_reset_table[j].data[1]); + } + } + ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); + for (int j = 0; j >= 0; j++) { + if (tas2781_post_reset_table[j].reg == 0) + break; + if (tas2781_post_reset_table[j].data[0] > 1) { + tasdevice_spi_dev_bulk_write(tas_dev, + tas2781_post_reset_table[j].reg, + &tas2781_post_reset_table[j].data[1], + tas2781_post_reset_table[j].data[0]); + } else { + tasdevice_spi_dev_write(tas_dev, + tas2781_post_reset_table[j].reg, + tas2781_post_reset_table[j].data[1]); + } + } +} + +static int tascodec_spi_init(struct tasdevice_priv *tas_priv, + void *codec, struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret; + + /* + * Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + guard(mutex)(&tas_priv->codec_lock); + + ret = scnprintf(tas_priv->rca_binaryname, + sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->index); + if (ret <= 0) { + dev_err(tas_priv->dev, "rca name err:0x%08x\n", ret); + return ret; + } + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + return ret; +} + +static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) +{ + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + tas_priv->cur_book = -1; + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + /* Store default registers address for calibration data. */ + tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); + tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); + tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); + tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); + tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + + mutex_init(&tas_priv->codec_lock); +} + +static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int val, ret; + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_update_bits(tas_priv, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret, val; + + /* Read the primary device */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + return ret; + } + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int val, ret; + + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "set digital vol err in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + return ret; + } + + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2781_read_acpi(struct tas2781_hda *tas_hda, + const char *hid, + int id) +{ + struct tasdevice_priv *p = tas_hda->priv; + struct acpi_device *adev; + struct device *physdev; + u32 values[HDA_MAX_COMPONENTS]; + const char *property; + size_t nval; + int ret, i; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, "Failed to find ACPI device: %s\n", hid); + return -ENODEV; + } + + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + tas_hda->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL */ + p->acpi_subsystem_id = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + + property = "ti,dev-index"; + ret = device_property_count_u32(physdev, property); + if (ret <= 0 || ret > ARRAY_SIZE(values)) { + ret = -EINVAL; + goto err; + } + nval = ret; + + ret = device_property_read_u32_array(physdev, property, values, nval); + if (ret) + goto err; + + p->index = U8_MAX; + for (i = 0; i < nval; i++) { + if (values[i] == id) { + p->index = i; + break; + } + } + if (p->index == U8_MAX) { + dev_dbg(p->dev, "No index found in %s\n", property); + ret = -ENODEV; + goto err; + } + + if (p->index == 0) { + /* All of amps share same RESET pin. */ + p->reset = devm_gpiod_get_index_optional(physdev, "reset", + p->index, GPIOD_OUT_LOW); + if (IS_ERR(p->reset)) { + dev_err_probe(p->dev, ret, "Failed on reset GPIO\n"); + goto err; + } + } + put_device(physdev); + + return 0; +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + acpi_dev_put(adev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + if (action == HDA_GEN_PCM_ACT_OPEN) { + pm_runtime_get_sync(dev); + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } else if (action == HDA_GEN_PCM_ACT_CLOSE) { + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->rcabin.ncfgs - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + if (tas_priv->rcabin.profile_cfg_id != val) { + tas_priv->rcabin.profile_cfg_id = val; + return 1; + } + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_priv->fmw->nr_programs - 1; + int val; + + val = clamp(nr_program, 0, max); + + if (tas_priv->cur_prog != val) { + tas_priv->cur_prog = val; + return 1; + } + + return 0; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->fmw->nr_configurations - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + + if (tas_priv->cur_conf != val) { + tas_priv->cur_conf = val; + return 1; + } + + return 0; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + * + * Return 0 if succeeded. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) { + change = false; + } else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 0", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 0", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 0", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 1", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 1", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 1", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl[] = { +{ + .name = "Speaker Profile Id - 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +{ + .name = "Speaker Profile Id - 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +}; +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl[] = { +{ + .name = "Speaker Program Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +{ + .name = "Speaker Program Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { +{ + .name = "Speaker Config Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +{ + .name = "Speaker Config Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + int j, rc; + + /* + * If no calibration data exist in tasdevice_priv *tas_priv, + * calibration apply will be ignored, and use default values + * in firmware binary, which was loaded during firmware download. + */ + if (tas_priv->cali_data[0] == 0) + return; + /* + * Calibration data was saved in tasdevice_priv *tas_priv as: + * unsigned int cali_data[CALIB_MAX]; + * and every data (in 4 bytes) will be saved in register which in + * book 0, and page number in page_array[], offset was saved in + * rgno_array[]. + */ + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + tas_priv->cali_reg_array[j], + (unsigned char *)&tas_priv->cali_data[j], 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + tas_priv->index, j, rc); + } +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; + unsigned int attr, crc, offset, *tmp_val; + struct tm *tm = &tas_priv->tm; + unsigned long total_sz = 0; + efi_status_t status; + + tas_priv->cali_data[0] = 0; + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); + if (status == EFI_BUFFER_TOO_SMALL) { + if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &total_sz, data); + } + if (status != EFI_SUCCESS) + return -EINVAL; + + tmp_val = (unsigned int *)data; + if (tmp_val[0] == 2781) { + /* + * New features were added in calibrated Data V3: + * 1. Added calibration registers address define in + * a node, marked as Device id == 0x80. + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and + * uniqie_id for multi-link solutions + * 2. Support flexible number of devices instead of + * fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2781, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } // if Device number is 0x80, mean it's + * calibration registers address. + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ + crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; + + if (crc != tmp_val[3 + tmp_val[1] * 6]) + return 0; + + time64_to_tm(tmp_val[2], 0, tm); + for (int j = 0; j < tmp_val[1]; j++) { + offset = j * 6 + 3; + if (tmp_val[offset] == tas_priv->index) { + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[offset + i + 1]; + } else if (tmp_val[offset] == + TASDEVICE_CALIBRATION_REG_ADDRESS) { + for (int i = 0; i < CALIB_MAX; i++) { + buf = &data[(offset + i + 1) * 4]; + tas_priv->cali_reg_array[i] = + TASDEVICE_REG(buf[1], buf[2], + buf[3]); + } + } + tas_priv->apply_calibration(tas_priv); + } + } else { + /* + * Calibration data is in V1 format. + * struct cali_data { + * char cali_data[20]; + * } + * + * struct { + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) + * } ueft; + */ + crc = crc32(~0, data, 84) ^ ~0; + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[tas_priv->index * 5 + i]; + tas_priv->apply_calibration(tas_priv); + } + } + + return 0; +} + +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct hda_codec *codec = tas_priv->codec; + int i, j, ret; + + pm_runtime_get_sync(tas_priv->dev); + guard(mutex)(&tas_priv->codec_lock); + + ret = tasdevice_spi_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + /* Add control one time only. */ + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl[tas_priv->index].name, ret); + goto out; + } + j = tas_priv->index * ARRAY_SIZE(tas2781_snd_controls) / 2; + for (i = 0; i < 3; i++) { + tas_hda->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i+j], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i+tas_priv->index*3].name, + ret); + goto out; + } + } + + tasdevice_spi_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", + codec->core.subsystem_id, tas_priv->index); + ret = tasdevice_spi_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + /* Add control one time only. */ + tas_hda->dsp_prog_ctl = + snd_ctl_new1(&tas2781_dsp_prog_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_hda->dsp_conf_ctl = + snd_ctl_new1(&tas2781_dsp_conf_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_spi_prmg_load(tas_priv, 0); + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; + + /* + * If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas_priv->save_calibration(tas_priv); + +out: + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_hda->priv->dev); + pm_runtime_put_autosuspend(tas_hda->priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + struct hda_codec *codec; + int ret; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (!comp) + return -EINVAL; + + if (comp->dev) + return -EBUSY; + + codec = parent->codec; + + pm_runtime_get_sync(dev); + + comp->dev = dev; + + strscpy(comp->name, dev_name(dev), sizeof(comp->name)); + + ret = tascodec_spi_init(tas_hda->priv, codec, THIS_MODULE, + tasdev_fw_ready); + if (!ret) + comp->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (comp && (comp->dev == dev)) { + comp->dev = NULL; + memset(comp->name, 0, sizeof(comp->name)); + comp->playback_hook = NULL; + } + + tas2781_hda_remove_controls(tas_hda); + + tasdevice_spi_config_info_remove(tas_hda->priv); + tasdevice_spi_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + component_del(tas_hda->priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_get_sync(tas_hda->priv->dev); + pm_runtime_disable(tas_hda->priv->dev); + + pm_runtime_put_noidle(tas_hda->priv->dev); + + mutex_destroy(&tas_hda->priv->codec_lock); +} + +static int tas2781_hda_spi_probe(struct spi_device *spi) +{ + struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; + const char *device_name; + int ret = 0; + + tas_hda = devm_kzalloc(&spi->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) + return -ENOMEM; + + spi->max_speed_hz = TAS2781_SPI_MAX_FREQ; + + tas_priv = devm_kzalloc(&spi->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + goto err; + tas_priv->dev = &spi->dev; + tas_hda->priv = tas_priv; + tas_priv->regmap = devm_regmap_init_spi(spi, &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate regmap: %d\n", + ret); + goto err; + } + if (strstr(dev_name(&spi->dev), "TXNW2781")) { + device_name = "TXNW2781"; + tas_priv->save_calibration = tas2781_save_calibration; + tas_priv->apply_calibration = tas2781_apply_calib; + } else { + goto err; + } + + tas_priv->irq = spi->irq; + dev_set_drvdata(&spi->dev, tas_hda); + ret = tas2781_read_acpi(tas_hda, device_name, + spi_get_chipselect(spi, 0)); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + tasdevice_spi_init(tas_priv); + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + tas2781_spi_reset(tas_priv); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component fail: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&spi->dev); + + return ret; +} + +static void tas2781_hda_spi_remove(struct spi_device *spi) +{ + tas2781_hda_remove(&spi->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + return 0; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret, val; + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + guard(mutex)(&tas_hda->priv->codec_lock); + + ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, + &val); + if (ret < 0) + return ret; + + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + tas_hda->priv->cur_prog = -1; + + tasdevice_spi_prmg_load(tas_hda->priv, 0); + tas_hda->priv->apply_calibration(tas_hda->priv); + + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct spi_device_id tas2781_hda_spi_id[] = { + { "tas2781-hda", }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TXNW2781", }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct spi_driver tas2781_hda_spi_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_spi_id, + .probe = tas2781_hda_spi_probe, + .remove = tas2781_hda_spi_remove, +}; +module_spi_driver(tas2781_hda_spi_driver); + +MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); +MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c new file mode 100644 index 000000000000..9e58e8d981e3 --- /dev/null +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -0,0 +1,2114 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <asm/unaligned.h> +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> + +#include "tas2781-spi.h" + +#define OFFSET_ERROR_BIT BIT(31) + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/* should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +struct blktyp_devidx_map { + unsigned char blktyp; + unsigned char dev_idx; +}; + +/* fixed m68k compiling issue: mapping table can save code field */ +static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0x81 }, + { PRE_DEVICE_A_1X, 0x81 }, + { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0x82 }, + { PRE_DEVICE_B_1X, 0x82 }, + { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0x83 }, + { PRE_DEVICE_C_1X, 0x83 }, + { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0x84 }, + { PRE_DEVICE_D_1X, 0x84 }, + { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, + { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, +}; + +static const struct blktyp_devidx_map ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, +}; + +static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES, 0x80 }, + { MAIN_DEVICE_A, 0x81 }, + { COEFF_DEVICE_A, 0xC1 }, + { PRE_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B, 0x82 }, + { COEFF_DEVICE_B, 0xC2 }, + { PRE_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C, 0x83 }, + { COEFF_DEVICE_C, 0xC3 }, + { PRE_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D, 0x84 }, + { COEFF_DEVICE_D, 0xC4 }, + { PRE_DEVICE_D, 0xC4 }, +}; +/* + * Device support different configurations for different scene, + * like voice, music, calibration, was write in regbin file. + * Will be stored into tas_priv after regbin was loaded. + */ +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* + * In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * DSP firmware. + */ + cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + return NULL; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if ((config_offset + 64) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto config_err; + } + config_offset += 64; + } + + if ((config_offset + 4) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto config_err; + } + + /* + * convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + /* + * Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(*bk_da), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto config_err; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + goto block_err; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + i++; + goto block_err; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + + return cfg_info; +block_err: + for (int j = 0; j < i; j++) + kfree(bk_da[j]); + kfree(bk_da); +config_err: + kfree(cfg_info); + return NULL; +} + +/* Regbin file parser function. */ +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + rca = &tas_priv->rcabin; + fw_hdr = &rca->fw_hdr; + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + buf = (unsigned char *)fmw->data; + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset++]; + fw_hdr->dev_family = buf[offset++]; + fw_hdr->reserve = buf[offset++]; + fw_hdr->ndev = buf[offset++]; + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", + fw_hdr->img_sz, total_config_sz, (int)offset); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -ENOMEM; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return ret; + } + offset += (int)fw_hdr->config_size[i]; + } + + return ret; +} + +/* fixed m68k compiling issue: mapping table can save code field */ +static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block) +{ + struct blktyp_devidx_map *p = + (struct blktyp_devidx_map *)non_ppc3_mapping_table; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + int i, n = ARRAY_SIZE(non_ppc3_mapping_table); + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; + n = ARRAY_SIZE(ppc3_tas2781_mapping_table); + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + p = (struct blktyp_devidx_map *)ppc3_mapping_table; + n = ARRAY_SIZE(ppc3_mapping_table); + } + + for (i = 0; i < n; i++) { + if (block->type == p[i].blktyp) { + dev_idx = p[i].dev_idx; + break; + } + } + + return dev_idx; +} + +/* Block parser function. */ +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host. + */ + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset++]; + block->pchksum = data[offset++]; + block->is_ychksum_present = data[offset++]; + block->ychksum = data[offset++]; + block->blk_size = get_unaligned_be32(&data[offset]); + offset += 4; + block->nr_subblocks = get_unaligned_be32(&data[offset]); + offset += 4; + + /* + * Fixed m68k compiling issue: + * 1. mapping table can save code field. + * 2. storing the dev_idx as a member of block can reduce unnecessary + * time and system resource comsumption of dev_idx mapping every + * time the block data writing to the dsp. + */ + block->dev_idx = map_dev_idx(tas_fmw, block); + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += block->blk_size; + + return offset; +} + +/* Data of block parser function. */ +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + img_data->nr_blk = get_unaligned_be32(&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data_kernel( + tas_fmw, blk, fmw, offset); + if (offset < 0) { + kfree(img_data->dev_blks); + return -EINVAL; + } + } + + return offset; +} + +/* Data of DSP program parser function. */ +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* skip 72 unused byts */ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* Data of DSP configurations parser function. */ +static int fw_parse_configuration_data_kernel(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* skip extra 16 bytes */ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* DSP firmware file header parser function for early PPC3 firmware binary. */ +static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_config *config; + struct tasdevice_prog *program; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + return -EINVAL; + } + offset += 2; + fw_hdr->device = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 2; + + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || + tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + return -EINVAL; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(*tas_fmw->programs), GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + program->prog_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); + offset += 4; + + /* + * The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) { + kfree(tas_fmw->programs); + return -ENOMEM; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &tas_fmw->configs[i]; + config->cfg_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + + return offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform single byte write to device. + */ +static int tasdevice_single_byte_wr(void *context, int dev_idx, + unsigned char *data, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int i, subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, "process_block: Out of boundary\n"); + return 0; + } + + for (i = 0; i < len; i++) { + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + dev_err(tas_priv->dev, + "process_block: single write error\n"); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 4; + } + + return subblk_offset; +} +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform multi-write to device. + */ +static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (len % 4) { + dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", + __func__, len); + subblk_offset |= OFFSET_ERROR_BIT; + } + + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &data[subblk_offset + 4], len); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += (len + 4); + + return subblk_offset; +} +/* Just delay for ms.*/ +static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned int sleep_time, subblk_offset = 2; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, "%s: delay Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + sleep_time = get_unaligned_be16(&data[2]) * 1000; + fsleep(sleep_time); + } + subblk_offset += 2; + + return subblk_offset; +} +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform bits update. + */ +static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int rc, subblk_offset = 2; + + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_update_bits(tas_priv, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: update_bits error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 6; + + return subblk_offset; +} +/* Data block process function. */ +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int blktyp = dev_idx & 0xC0, subblk_offset; + unsigned char subblk_typ = data[1]; + + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: + subblk_offset = tasdevice_single_byte_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_BURST: + subblk_offset = tasdevice_burst_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_DELAY: + subblk_offset = tasdevice_delay(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_FIELD_W: + subblk_offset = tasdevice_field_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + default: + subblk_offset = 2; + break; + } + if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + } else + tas_priv->cur_conf = -1; + } + subblk_offset &= ~OFFSET_ERROR_BIT; + + return subblk_offset; +} +/* + * Device support different configurations for different scene, + * this function was used for choose different config. + */ +void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = pContext; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + unsigned int j, k; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < blk_data[j]->n_subblks; k++) { + tas_priv->is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} +/* Block process function. */ +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + const unsigned int blk_size = block->blk_size; + unsigned char *data = block->data; + unsigned int i, length; + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + block->dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} +/* DSP firmware file header parser function. */ +static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, + struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + offset += len; + + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + return -EINVAL; + } + offset += 4; + + fw_hdr->device = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 4; + fw_hdr->ndev = 1; + + return offset; +} + +/* DSP firmware file header parser function for size variabled header. */ +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + + return offset; +} + +/* DSP firmware file block parser function. */ +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + return -EINVAL; + } + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + return -EINVAL; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = get_unaligned_be32(&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += n; + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i, n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + return -EINVAL; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + return -EINVAL; + } + offset += n; + img_data->nr_blk = get_unaligned_be16(&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(*img_data->dev_blks), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) + return -EINVAL; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /* Not error in calibration Data file, return directly */ + dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + return offset; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), + GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &tas_fmw->programs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, + offset); + if (offset < 0) + return offset; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /* Not error for calibration Data file, return directly */ + return offset; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) + return -ENOMEM; + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + offset += n; + offset = fw_parse_data(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* yram5 page check. */ +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +/* DSP firmware yram block check. */ +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) { + in = check_inpage_yram_rg(cd, reg, len); + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) + in = check_inpage_yram_bk1(cd, page, reg, len); + else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} +/* yram4 page check. */ +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} +/* yram page check. */ +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (!in) + in = check_inblock_yram(cd, book, page, reg, len); + + return in; +} + +/* Checksum for data block. */ +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0, i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (len == 4)) { + /* DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_spi_dev_bulk_read(tasdevice, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) >= + TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) <= + (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + /* DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +/* Checksum for single register. */ +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + /* DSP swap command, pass */ + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + return 0; + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + return 0; + ret = tasdevice_spi_dev_read(tasdevice, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + return ret; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->err_code |= ERROR_YRAM_CRCCHK; + return -EAGAIN; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + + return ret; +} + +/* Block type check. */ +static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || + (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || + (type == MAIN_DEVICE_D)) + p->cur_prog = -1; + else + p->cur_conf = -1; +} +/* Checksum for data bytes. */ +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, book, page, reg, val); + + if (ret > 0) { + *crc_chksum += ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, tas_priv); + +end: + return ret; +} +/* Multi-data byte write. */ +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + + return ret; +} +/* Checksum for block. */ +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); + set_err_prg_cfg(block->type, tas_priv); + return ret; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, tas_priv); + } else { + tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; + } + + return ret; +} +/* Firmware block load function. */ +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int sleep_time, len, nr_cmds; + unsigned char offset, book, page, val; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_CHECKSUM, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /* Single byte write */ + if (offset <= 0x7F) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + break; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /* sleep command */ + if (offset == 0x81) { + /* book -- data[0] page -- data[1] */ + sleep_time = ((book << 8) + page)*1000; + fsleep(sleep_time); + continue; + } + /* Multiple bytes write */ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /* skip current blk */ + break; + } + + return ret; +} +/* Firmware block load function. */ +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int ret = 0; + + block->nr_retry = 6; + if (tas_priv->is_loading == false) + return 0; + ret = tasdev_load_blk(tas_priv, block); + if (ret < 0) + dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); + + return ret; +} + +/* + * Select firmware binary parser & load callback functions by ppc3 version + * and firmware binary version. + */ +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} +/* Calibration block load function. */ +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware binary file header parser function. */ +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); + offset += 72; + +out: + return offset; +} + +/* DSP firmware binary file parser function. */ +static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + return -EINVAL; + } + + tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) + return -EINVAL; + + fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + case 0x503: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + return ret; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + + return ret; +} +/* DSP firmware binary file parser function. */ +int tasdevice_spi_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + return ret; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + + return ret; +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &prog->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +/* DSP firmware config block data remove function. */ +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &cfg->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +/* DSP firmware config remove function. */ +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +/* DSP firmware remove function. */ +void tasdevice_spi_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = context; + + if (!tas_dev->fmw) + return; + + if (tas_dev->fmw->programs) + tasdev_dsp_prog_remove(tas_dev->fmw->programs, + tas_dev->fmw->nr_programs); + if (tas_dev->fmw->configs) + tasdev_dsp_cfg_remove(tas_dev->fmw->configs, + tas_dev->fmw->nr_configurations); + kfree(tas_dev->fmw); + tas_dev->fmw = NULL; +} + +/* DSP firmware calibration data remove function. */ +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &tas_fmw->calibrations[i]; + if (!calibration) + continue; + + if (!calibration->dev_data.dev_blks) + continue; + + for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { + block = &calibration->dev_data.dev_blks[blks]; + if (!block) + continue; + kfree(block->data); + } + kfree(calibration->dev_data.dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +/* Calibration data from firmware remove function. */ +void tasdevice_spi_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + + if (tas_priv->cali_data_fmw) { + tas2781_clear_calfirmware(tas_priv->cali_data_fmw); + tas_priv->cali_data_fmw = NULL; + } +} + +/* Configuration remove function. */ +void tasdevice_spi_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **ci = rca->cfg_info; + unsigned int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} + +/* DSP firmware program block data load function. */ +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware program load function. */ +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (rca_conf_no >= rca->ncfgs || + rca_conf_no < 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + return 0; + } + + if (prm_no >= 0 && + (tas_priv->cur_prog != prm_no || + tas_priv->force_fwload_status)) { + tas_priv->cur_conf = -1; + tas_priv->is_loading = true; + prog_status++; + } + + if (prog_status) { + program = &tas_fmw->programs[prm_no]; + tasdevice_load_data(tas_priv, &program->dev_data); + if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &cal->dev_data); + } + tas_priv->cur_prog = prm_no; + } + + } + + if (cfg_no && + (cfg_no != tas_priv->cur_conf) && + (tas_priv->is_loaderr == false)) { + status++; + tas_priv->is_loading = true; + } else { + tas_priv->is_loading = false; + } + if (status) { + conf = &tas_fmw->configs[cfg_no]; + status = 0; + tasdevice_load_data(tas_priv, &conf->dev_data); + if (tas_priv->is_loaderr == true) { + status |= 1 << 4; + } else if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + tas_priv->cur_conf = cfg_no; + } + } else { + dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", + __func__, cfg_no); + } + + return prog_status; +} + +/* DSP firmware program load interface function. */ +int tasdevice_spi_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (prm_no >= 0) { + tas_priv->cur_conf = 0; + tas_priv->is_loading = true; + prog_status++; + } + if (prog_status) { + program = &tas_fmw->programs[prm_no]; + tasdevice_load_data(tas_priv, &program->dev_data); + tas_priv->cur_prog = prm_no; + } + if (prog_status) { + conf = &tas_fmw->configs[tas_priv->cur_conf]; + tasdevice_load_data(tas_priv, &conf->dev_data); + } + + return prog_status; +} + +/* DSP firmware program switch interface function. */ +void tasdevice_spi_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = context; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else { + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } +} -- 2.40.1

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