From 965cc040bf0698e81b0c0aef359ae650b42b428e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 20 May 2024 16:19:56 +0200 Subject: ASoC: Constify channel mapping array arguments in set_channel_map() There is no need for implementations of DAI set_channel_map() to modify contents of passed arrays with actual channel mapping. Additionally, the caller keeps full ownership of the array. Constify these pointer arguments so the code will be safer and easier to read (documenting the caller's ownership). Acked-by: Charles Keepax Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240520-asoc-x1e80100-4-channel-mapping-v4-1-f657159b4aad@linaro.org Signed-off-by: Mark Brown --- include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h index bb70782d15d0..43c6a9ef8d9f 100644 --- a/include/sound/cs35l41.h +++ b/include/sound/cs35l41.h @@ -896,8 +896,8 @@ int cs35l41_test_key_lock(struct device *dev, struct regmap *regmap); int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap); int cs35l41_register_errata_patch(struct device *dev, struct regmap *reg, unsigned int reg_revid); int cs35l41_set_channels(struct device *dev, struct regmap *reg, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot); int cs35l41_gpio_config(struct regmap *regmap, struct cs35l41_hw_cfg *hw_cfg); void cs35l41_configure_cs_dsp(struct device *dev, struct regmap *reg, struct cs_dsp *dsp); int cs35l41_set_cspl_mbox_cmd(struct device *dev, struct regmap *regmap, diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index adcd8719d343..15ef268c9845 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -188,8 +188,8 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -305,8 +305,8 @@ struct snd_soc_dai_ops { unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int (*set_channel_map)(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot); int (*get_channel_map)(struct snd_soc_dai *dai, unsigned int *tx_num, unsigned int *tx_slot, unsigned int *rx_num, unsigned int *rx_slot); -- cgit v1.2.3 From 734447685ecc7c6328e40cb1bd4aaeeac03c1413 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Mon, 13 May 2024 19:37:20 +0200 Subject: ASoC: topology: Constify an argument of snd_soc_tplg_component_load() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_soc_tplg_component_load() does not modify its "*ops" argument. It only read some values and stores it in "soc_tplg.ops". This argument and the ops field in "struct soc_tplg" can be made const. Signed-off-by: Christophe JAILLET Reviewed-by: Amadeusz Sławiński Link: https://msgid.link/r/f2f983e791d7f941a95556bb147f426a345d84d4.1715526069.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f055c6917f6c..1eedd203ac29 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -178,7 +178,7 @@ static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) /* Dynamic Object loading and removal for component drivers */ int snd_soc_tplg_component_load(struct snd_soc_component *comp, - struct snd_soc_tplg_ops *ops, const struct firmware *fw); + const struct snd_soc_tplg_ops *ops, const struct firmware *fw); int snd_soc_tplg_component_remove(struct snd_soc_component *comp); /* Binds event handlers to dynamic widgets */ -- cgit v1.2.3 From 8cb3aeebcb86088e30232277c9bee9054837442b Mon Sep 17 00:00:00 2001 From: Mohan Kumar Date: Mon, 27 May 2024 12:56:06 +0000 Subject: ASoC: simple-card-utils: Split simple_fixup_sample_fmt func Split the simple_fixup_sample_fmt() into two functions by adding one more function named simple_util_get_sample_fmt() to return the sample format value. This is useful for drivers that wish to simply get the sample format without setting the mask. Signed-off-by: Mohan Kumar Signed-off-by: Sameer Pujar Link: https://msgid.link/r/20240527125608.2461300-2-spujar@nvidia.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index ad67957b7b48..2c2279d082ec 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -174,6 +174,8 @@ void simple_util_parse_convert(struct device_node *np, char *prefix, struct simple_util_data *data); bool simple_util_is_convert_required(const struct simple_util_data *data); +int simple_util_get_sample_fmt(struct simple_util_data *data); + int simple_util_parse_routing(struct snd_soc_card *card, char *prefix); int simple_util_parse_widgets(struct snd_soc_card *card, -- cgit v1.2.3 From 1deba6e24c221c61c6eab8656a53f8c17035932b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 May 2024 14:35:47 -0500 Subject: ASoC: SOF: sof-audio: add sof_dai_get_tdm_slots function An new interface, sof_dai_get_tdm_slots(), is added for machine driver to get tdm slot number from topology. The dai_get_param() callback needs to support new parameter type SOF_DAI_PARAM_INTEL_SSP_TDM_SLOTS by returning the tdm slot number of specific SSP port. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240527193552.165567-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/sof.h b/include/sound/sof.h index ec6c30d54592..64fd5504cb2b 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -173,5 +173,6 @@ struct sof_dev_desc { int sof_dai_get_mclk(struct snd_soc_pcm_runtime *rtd); int sof_dai_get_bclk(struct snd_soc_pcm_runtime *rtd); +int sof_dai_get_tdm_slots(struct snd_soc_pcm_runtime *rtd); #endif -- cgit v1.2.3 From 45919c28134519080a85a5fb66d0f65955ef7572 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 28 May 2024 05:05:14 +0000 Subject: ASoC: simple-card-utils: remove both playback/capture_only check soc-pcm.c :: soc_get_playback_capture() will indicate error if both playback_only / capture_only were true. Thus, graph_util_parse_link_direction() which setup playback_only / capture_only don't need to check it. And, its return value is not used on existing driver. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87a5kah6gm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 2c2279d082ec..0a6435ac5c5f 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -197,7 +197,7 @@ int graph_util_is_ports0(struct device_node *port); int graph_util_parse_dai(struct device *dev, struct device_node *ep, struct snd_soc_dai_link_component *dlc, int *is_single_link); -int graph_util_parse_link_direction(struct device_node *np, +void graph_util_parse_link_direction(struct device_node *np, bool *is_playback_only, bool *is_capture_only); #ifdef DEBUG -- cgit v1.2.3 From 0b7e448119428e1dcb854abb5855f66966fb82dc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 28 May 2024 14:29:33 -0500 Subject: ACPI: utils: introduce acpi_get_local_u64_address() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ACPI _ADR is a 64-bit value. We changed the definitions in commit ca6f998cf9a2 ("ACPI: bus: change _ADR representation to 64 bits") but some helpers still assume the value is a 32-bit value. This patch adds a new helper to extract the full 64-bits. The existing 32-bit helper is kept for backwards-compatibility and cases where the _ADR is known to fit in a 32-bit value. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Acked-by: Rafael J. Wysocki Reviewed-by: Takashi Iwai Reviewed-by: Andy Shevchenko Link: https://lore.kernel.org/r/20240528192936.16180-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/linux/acpi.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/acpi.h b/include/linux/acpi.h index 28c3fb2bef0d..65e7177bcb02 100644 --- a/include/linux/acpi.h +++ b/include/linux/acpi.h @@ -761,6 +761,7 @@ static inline u64 acpi_arch_get_root_pointer(void) } #endif +int acpi_get_local_u64_address(acpi_handle handle, u64 *addr); int acpi_get_local_address(acpi_handle handle, u32 *addr); const char *acpi_get_subsystem_id(acpi_handle handle); -- cgit v1.2.3 From 163f10b2935362f0e8ef8d7fadd0b5aa33e9130f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 12 Jun 2024 08:47:07 +0200 Subject: PCI: Add INTEL_HDA_PTL to pci_ids.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit More PCI ids for Intel audio. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Acked-by: Bjorn Helgaas Link: https://lore.kernel.org/r/20240612064709.51141-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 942a587bb97e..0168c6a60148 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -3112,6 +3112,7 @@ #define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828 #define PCI_DEVICE_ID_INTEL_S21152BB 0xb152 #define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7 +#define PCI_DEVICE_ID_INTEL_HDA_PTL 0xe428 #define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8 #define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8 -- cgit v1.2.3 From 1c75adb22d49ca9389333ca5e6939052a7203111 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Thu, 13 Jun 2024 22:59:09 +0200 Subject: ASoC: SOF: mediatek: Constify struct mtk_adsp_ipc_ops 'struct mtk_adsp_ipc_ops' is not modified in these drivers. Constifying this structure moves some data to a read-only section, so increase overall security. In order to do it, "struct mtk_adsp_ipc" also needs to be adjusted to this new const qualifier. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 15533 2383 0 17916 45fc sound/soc/sof/mediatek/mt8195/mt8195.o After: ===== text data bss dec hex filename 15557 2367 0 17924 4604 sound/soc/sof/mediatek/mt8195/mt8195.o Signed-off-by: Christophe JAILLET Link: https://msgid.link/r/a45d6b2b5ec040ea0fc78fca662c2dca3f13a49f.1718312321.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- include/linux/firmware/mediatek/mtk-adsp-ipc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/linux/firmware/mediatek/mtk-adsp-ipc.h b/include/linux/firmware/mediatek/mtk-adsp-ipc.h index 5b1d16fa3f56..6e86799a7dc4 100644 --- a/include/linux/firmware/mediatek/mtk-adsp-ipc.h +++ b/include/linux/firmware/mediatek/mtk-adsp-ipc.h @@ -40,7 +40,7 @@ struct mtk_adsp_chan { struct mtk_adsp_ipc { struct mtk_adsp_chan chans[MTK_ADSP_MBOX_NUM]; struct device *dev; - struct mtk_adsp_ipc_ops *ops; + const struct mtk_adsp_ipc_ops *ops; void *private_data; }; -- cgit v1.2.3 From d4a7d067e061c95c6387cf537258082074a4d299 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 14:57:34 +0200 Subject: ASoC: soc-dai.h: Constify DAI ops auto_selectable_formats The core ASoC code does not modify contents of the 'auto_selectable_formats' array passed in 'struct snd_soc_dai_ops', so make it const for code safety. Reviewed-by: Herve Codina Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617125735.582963-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 15ef268c9845..279223c4ef5e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -361,7 +361,7 @@ struct snd_soc_dai_ops { * see * snd_soc_dai_get_fmt() */ - u64 *auto_selectable_formats; + const u64 *auto_selectable_formats; int num_auto_selectable_formats; /* probe ordering - for components with runtime dependencies */ -- cgit v1.2.3 From 2fbafecb0f05818e25f6c926c6f9ad9ef597429c Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 15:03:19 +0200 Subject: ASoC: Constify of_phandle_args in snd_soc_dai_driver ASoC core code does not modify contents of 'of_phandle_args' in 'struct snd_soc_dai_driver', so the pointer can be made as a pointer to const. This makes code safer, serves as clear annotation of core's intentions and allows putting pointed structures in rodata (if ever applicable). Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-1-8004f346ee38@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 15ef268c9845..e6c61c79c483 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -413,7 +413,7 @@ struct snd_soc_dai_driver { unsigned int id; unsigned int base; struct snd_soc_dobj dobj; - struct of_phandle_args *dai_args; + const struct of_phandle_args *dai_args; /* ops */ const struct snd_soc_dai_ops *ops; -- cgit v1.2.3 From 020b37d06f97de289940805bc821190d5858eda0 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 15:03:20 +0200 Subject: ASoC: Constify of_phandle_args in snd_soc_dai_link_component ASoC core code does not modify contents of 'of_phandle_args' in 'struct snd_soc_dai_link_component', so the pointer can be made as a pointer to const. This makes code safer, serves as clear annotation of core's intentions and allows putting pointed structures in rodata (if ever applicable). Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-2-8004f346ee38@linaro.org Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 33671437ee89..f02a51694ab4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -675,7 +675,7 @@ struct snd_soc_dai_link_component { const char *name; struct device_node *of_node; const char *dai_name; - struct of_phandle_args *dai_args; + const struct of_phandle_args *dai_args; }; /* -- cgit v1.2.3 From f3ac3da7e4d0957b3402fb31a4ca480e739e086f Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 15:03:21 +0200 Subject: ASoC: Constify passed data to core function Several ASoC functions receive pointers to data which is not modified, e.g. pointers to 'snd_soc_dai', 'snd_soc_pcm_runtime', 'snd_pcm_hw_params' and 'snd_soc_dai_link'. All these pointers can be made as a pointer to const. This makes code safer, serves as clear annotation of function's intentions (no ownership passed to the function, no modifications) and allows putting pointed structures in rodata (if ever applicable). Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-3-8004f346ee38@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 18 ++++++++++-------- include/sound/soc.h | 17 +++++++++-------- 2 files changed, 19 insertions(+), 16 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e6c61c79c483..3c45b418270e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -180,8 +180,8 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ -int snd_soc_dai_get_fmt_max_priority(struct snd_soc_pcm_runtime *rtd); -u64 snd_soc_dai_get_fmt(struct snd_soc_dai *dai, int priority); +int snd_soc_dai_get_fmt_max_priority(const struct snd_soc_pcm_runtime *rtd); +u64 snd_soc_dai_get_fmt(const struct snd_soc_dai *dai, int priority); int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -202,7 +202,7 @@ int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, unsigned int *tx_num, unsigned int *tx_slot, unsigned int *rx_num, unsigned int *rx_slot); -int snd_soc_dai_is_dummy(struct snd_soc_dai *dai); +int snd_soc_dai_is_dummy(const struct snd_soc_dai *dai); int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, @@ -218,7 +218,7 @@ void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); -bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream); +bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); @@ -232,7 +232,7 @@ static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai, { snd_soc_dai_action(dai, stream, -1); } -int snd_soc_dai_active(struct snd_soc_dai *dai); +int snd_soc_dai_active(const struct snd_soc_dai *dai); int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order); int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order); @@ -271,7 +271,7 @@ int snd_soc_dai_compr_get_metadata(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata); -const char *snd_soc_dai_name_get(struct snd_soc_dai *dai); +const char *snd_soc_dai_name_get(const struct snd_soc_dai *dai); struct snd_soc_dai_ops { /* DAI driver callbacks */ @@ -518,7 +518,8 @@ static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *play snd_soc_dai_dma_data_set_capture(dai, capture); } -static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream) +static inline unsigned int snd_soc_dai_tdm_mask_get(const struct snd_soc_dai *dai, + int stream) { return dai->stream[stream].tdm_mask; } @@ -529,7 +530,8 @@ static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream, dai->stream[stream].tdm_mask = tdm_mask; } -static inline unsigned int snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream) +static inline unsigned int snd_soc_dai_stream_active(const struct snd_soc_dai *dai, + int stream) { /* see snd_soc_dai_action() for setup */ return dai->stream[stream].active; diff --git a/include/sound/soc.h b/include/sound/soc.h index f02a51694ab4..a8e66bbf932b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -534,10 +534,10 @@ static inline int snd_soc_set_dmi_name(struct snd_soc_card *card, /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); -int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); +int snd_soc_params_to_frame_size(const struct snd_pcm_hw_params *params); int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots); -int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms); -int snd_soc_tdm_params_to_bclk(struct snd_pcm_hw_params *params, +int snd_soc_params_to_bclk(const struct snd_pcm_hw_params *parms); +int snd_soc_tdm_params_to_bclk(const struct snd_pcm_hw_params *params, int tdm_width, int tdm_slots, int slot_multiple); /* set runtime hw params */ @@ -837,7 +837,8 @@ struct snd_soc_dai_link { #endif }; -static inline int snd_soc_link_num_ch_map(struct snd_soc_dai_link *link) { +static inline int snd_soc_link_num_ch_map(const struct snd_soc_dai_link *link) +{ return max(link->num_cpus, link->num_codecs); } @@ -1299,7 +1300,7 @@ struct soc_enum { #endif }; -static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) +static inline bool snd_soc_volsw_is_stereo(const struct soc_mixer_control *mc) { if (mc->reg == mc->rreg && mc->shift == mc->rshift) return false; @@ -1311,7 +1312,7 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) return true; } -static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e, +static inline unsigned int snd_soc_enum_val_to_item(const struct soc_enum *e, unsigned int val) { unsigned int i; @@ -1326,7 +1327,7 @@ static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e, return 0; } -static inline unsigned int snd_soc_enum_item_to_val(struct soc_enum *e, +static inline unsigned int snd_soc_enum_item_to_val(const struct soc_enum *e, unsigned int item) { if (!e->values) @@ -1401,7 +1402,7 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, snd_soc_daifmt_clock_provider_from_bitmap( \ snd_soc_daifmt_parse_clock_provider_as_bitmap(np, prefix)) -int snd_soc_get_stream_cpu(struct snd_soc_dai_link *dai_link, int stream); +int snd_soc_get_stream_cpu(const struct snd_soc_dai_link *dai_link, int stream); int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_component *dlc); int snd_soc_of_get_dlc(struct device_node *of_node, -- cgit v1.2.3 From 785d64c4941221044940ab199e6625af17296470 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 15:03:22 +0200 Subject: ASoC: Constify DAI passed to get_channel_map get_channel_map() is supposed to obtain map of channels without modifying the state of the given DAI, so make the pointer to 'struct snd_soc_dai' as pointing to const. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-4-8004f346ee38@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 3c45b418270e..f22969298de1 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -198,7 +198,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); -int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai, +int snd_soc_dai_get_channel_map(const struct snd_soc_dai *dai, unsigned int *tx_num, unsigned int *tx_slot, unsigned int *rx_num, unsigned int *rx_slot); @@ -307,7 +307,7 @@ struct snd_soc_dai_ops { int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, const unsigned int *tx_slot, unsigned int rx_num, const unsigned int *rx_slot); - int (*get_channel_map)(struct snd_soc_dai *dai, + int (*get_channel_map)(const struct snd_soc_dai *dai, unsigned int *tx_num, unsigned int *tx_slot, unsigned int *rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); -- cgit v1.2.3 From de267e7a6ea8e6fa29af2287adfc9fc9d87e6dc9 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 17 Jun 2024 15:03:23 +0200 Subject: ASoC: Constify return of snd_soc_dai_get_pcm_stream() Returned 'struct snd_soc_pcm_stream' by snd_soc_dai_get_pcm_stream() is not modified by the users, so it can be changed as pointer to const. This is a necessary step towards making the 'dai->driver' a pointer to const. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240617-n-asoc-const-auto-selectable-formats-v1-5-8004f346ee38@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index f22969298de1..bd249710d716 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -473,7 +473,7 @@ struct snd_soc_dai { unsigned int probed:1; }; -static inline struct snd_soc_pcm_stream * +static inline const struct snd_soc_pcm_stream * snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) { return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? -- cgit v1.2.3 From 9f774c757e3fb2ac32dc4377e8f21f3364a8df81 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Fri, 14 Jun 2024 21:36:45 +0800 Subject: ASoc: tas2781: Enable RCA-based playback without DSP firmware download In only loading RCA (Reconfigurable Architecture) binary case, no DSP program will be working inside tas2563/tas2781, that is dsp-bypass mode, do not support speaker protection, or audio acoustic algorithms in this mode. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240614133646.910-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781-dsp.h | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..3cda9da14f6d 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -117,10 +117,17 @@ struct tasdevice_fw { struct device *dev; }; -enum tasdevice_dsp_fw_state { - TASDEVICE_DSP_FW_NONE = 0, +enum tasdevice_fw_state { + /* Driver in startup mode, not load any firmware. */ TASDEVICE_DSP_FW_PENDING, + /* DSP firmware in the system, but parsing error. */ TASDEVICE_DSP_FW_FAIL, + /* + * Only RCA (Reconfigurable Architecture) firmware load + * successfully. + */ + TASDEVICE_RCA_FW_OK, + /* Both RCA and DSP firmware load successfully. */ TASDEVICE_DSP_FW_ALL_OK, }; -- cgit v1.2.3 From fe1ff61487ace69cd4e680e36169c90667eb9624 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Wed, 19 Jun 2024 05:53:42 +0000 Subject: ASoC: rt1318: Add RT1318 audio amplifier driver This is the initial i2s-based amplifier driver for rt1318. Signed-off-by: Jack Yu Link: https://msgid.link/r/b3055442ce6d4994aa01aa1fad6ba1fe@realtek.com Signed-off-by: Mark Brown --- include/sound/rt1318.h | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) create mode 100644 include/sound/rt1318.h (limited to 'include') diff --git a/include/sound/rt1318.h b/include/sound/rt1318.h new file mode 100644 index 000000000000..fe6bff06036c --- /dev/null +++ b/include/sound/rt1318.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1318.h -- Platform data for RT1318 + * + * Copyright 2024 Realtek Semiconductor Corp. + */ + +#ifndef __LINUX_SND_RT1318_H +#define __LINUX_SND_RT1318_H + +struct rt1318_platform_data { + unsigned int init_r0_l; + unsigned int init_r0_r; +}; + +#endif -- cgit v1.2.3 From 00dd4d86ed908e70d912a96ad91d1248ff055b62 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Fri, 21 Jun 2024 21:23:07 +0800 Subject: ASoc: tas2781: Add name_prefix as the prefix name of firmwares and kcontrol to support corresponding TAS2563/TAS2781s Add name_prefix as the prefix name of firmwares and kcontrol to support corresponding TAS2563/TAS2781s. name_prefix is not mandatory. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240621132309.564-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..cd8ce522b78e 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -108,6 +108,7 @@ struct tasdevice_priv { unsigned char coef_binaryname[64]; unsigned char rca_binaryname[64]; unsigned char dev_name[32]; + const char *name_prefix; unsigned char ndev; unsigned int magic_num; unsigned int chip_id; -- cgit v1.2.3 From fafc20ded3f4659873c83c2af6d389983d480994 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 12 Jun 2024 06:02:26 +0000 Subject: ASoC: audio-graph-port: add link-trigger-order Sound Card need to consider/adjust HW control ordering based on the combination of CPU/Codec. The controlling feature is already supported on ASoC, but Simple Audio Card / Audio Graph Card still not support it. Let's support it. Cc: Maxim Kochetkov Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87sexizojx.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/dt-bindings/sound/audio-graph.h | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 include/dt-bindings/sound/audio-graph.h (limited to 'include') diff --git a/include/dt-bindings/sound/audio-graph.h b/include/dt-bindings/sound/audio-graph.h new file mode 100644 index 000000000000..bdb70c6b7332 --- /dev/null +++ b/include/dt-bindings/sound/audio-graph.h @@ -0,0 +1,26 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * audio-graph.h + * + * Copyright (c) 2024 Kuninori Morimoto + */ +#ifndef __AUDIO_GRAPH_H +#define __AUDIO_GRAPH_H + +/* + * used in + * link-trigger-order + * link-trigger-order-start + * link-trigger-order-stop + * + * default is + * link-trigger-order = ; + */ +#define SND_SOC_TRIGGER_LINK 0 +#define SND_SOC_TRIGGER_COMPONENT 1 +#define SND_SOC_TRIGGER_DAI 2 +#define SND_SOC_TRIGGER_SIZE 3 /* shoud be last */ + +#endif /* __AUDIO_GRAPH_H */ -- cgit v1.2.3 From 5d9cacdccf17bd33dac3ea378324650159c2a863 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 12 Jun 2024 06:02:33 +0000 Subject: ASoC: simple-card-utils: add link-trigger-order support Some Sound Card might need special trigger ordering which is based on CPU/Codec connection. It is already supported on ASoC, but Simple Audio Card / Audio Graph Card still not support it. Let's support it. Cc: Maxim Kochetkov Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87r0d2zojq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0a6435ac5c5f..3360d9eab068 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -199,6 +199,10 @@ int graph_util_parse_dai(struct device *dev, struct device_node *ep, void graph_util_parse_link_direction(struct device_node *np, bool *is_playback_only, bool *is_capture_only); +void graph_util_parse_trigger_order(struct simple_util_priv *priv, + struct device_node *np, + enum snd_soc_trigger_order *trigger_start, + enum snd_soc_trigger_order *trigger_stop); #ifdef DEBUG static inline void simple_util_debug_dai(struct simple_util_priv *priv, -- cgit v1.2.3 From 5d7e328e20b3d2bd3e1e8bea7a868ab8892aeed1 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 1 Jul 2024 11:44:42 +0100 Subject: ASoC: cs35l56: Revert support for dual-ownership of ASP registers This patch reverts a series of commits that allowed for the ASP registers to be owned by either the driver or the firmware. Nothing currently depends on the functionality that is being reverted, so it is safe to remove. The commits being reverted are (last 3 are bugfixes to the first 2): commit 72a77d7631c6 ("ASoC: cs35l56: Fix to ensure ASP1 registers match cache") commit 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") commit 4703b014f28b ("ASoC: cs35l56: fix reversed if statement in cs35l56_dspwait_asp1tx_put()") commit c14f09f010cc ("ASoC: cs35l56: Fix deadlock in ASP1 mixer register initialization") commit dfd2ffb37399 ("ASoC: cs35l56: Prevent overwriting firmware ASP config") These reverts have been squashed into a single commit because there would be no reason to revert only some of them (which would just reintroduce bugs). The changes introduced by the commits were well-intentioned but somewhat misguided. ACPI does not provide any information about how audio hardware is linked together, so that information has to be hardcoded into drivers. On Windows the firmware is customized to statically setup appropriate configuration of the audio links, and the intent of the commits was to re-use this information if the Linux host drivers aren't taking control of the ASP. This would avoid having to hardcode the ASP config into the machine driver on some systems. However, this added complexity and race conditions into the driver. It also complicates implementation of new code. The only case where the ASP is used but the host is not taking ownership is when CS35L56 is used in SoundWire mode with the ASP as a reference audio interconnect. But even in that case it's not necessarily required even if the firmware initialized it. Typically it is used to avoid the host SDCA drivers having to be capable of aggregating capture paths from multiple SoundWire peripherals. But the SOF SoundWire support is capable of doing that aggregation. Reverting all these commits significantly simplifies the driver. Let's just use the normal Linux mechanisms of the machine driver and ALSA controls to set things up instead of trying to use the firmware to do use-case setup. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20240701104444.172556-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 1a3c6f66f620..2e19722989f7 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -267,13 +267,18 @@ struct cs35l56_base { bool fw_patched; bool secured; bool can_hibernate; - bool fw_owns_asp1; bool cal_data_valid; s8 cal_index; struct cirrus_amp_cal_data cal_data; struct gpio_desc *reset_gpio; }; +/* Temporary to avoid a build break with the HDA driver */ +static inline int cs35l56_force_sync_asp1_registers_from_cache(struct cs35l56_base *cs35l56_base) +{ + return 0; +} + extern struct regmap_config cs35l56_regmap_i2c; extern struct regmap_config cs35l56_regmap_spi; extern struct regmap_config cs35l56_regmap_sdw; @@ -284,8 +289,6 @@ extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC]; extern const unsigned int cs35l56_tx_input_values[CS35L56_NUM_INPUT_SRC]; int cs35l56_set_patch(struct cs35l56_base *cs35l56_base); -int cs35l56_init_asp1_regs_for_driver_control(struct cs35l56_base *cs35l56_base); -int cs35l56_force_sync_asp1_registers_from_cache(struct cs35l56_base *cs35l56_base); int cs35l56_mbox_send(struct cs35l56_base *cs35l56_base, unsigned int command); int cs35l56_firmware_shutdown(struct cs35l56_base *cs35l56_base); int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base); -- cgit v1.2.3 From e2996141d6db0d8b353e1c221a37c8e1be109d4a Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 1 Jul 2024 11:44:43 +0100 Subject: ASoC: cs35l56: Remove support for A1 silicon No product was ever released with A1 silicon so there is no need for the driver to include support for it. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20240701104444.172556-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 2e19722989f7..642ef690ebc2 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -80,9 +80,7 @@ #define CS35L56_DSP1_AHBM_WINDOW_DEBUG_1 0x25E2044 #define CS35L56_DSP1_XMEM_UNPACKED24_0 0x2800000 #define CS35L56_DSP1_FW_VER 0x2800010 -#define CS35L56_DSP1_HALO_STATE_A1 0x2801E58 #define CS35L56_DSP1_HALO_STATE 0x28021E0 -#define CS35L56_DSP1_PM_CUR_STATE_A1 0x2804000 #define CS35L56_DSP1_PM_CUR_STATE 0x2804308 #define CS35L56_DSP1_XMEM_UNPACKED24_8191 0x2807FFC #define CS35L56_DSP1_CORE_BASE 0x2B80000 -- cgit v1.2.3 From f05c1ffc274516ef101d2e0f860bcb9b08c6c622 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 25 Jun 2024 19:25:46 +0200 Subject: ALSA: pcm: reinvent the stream synchronization ID API Until the commit e11f0f90a626 ("ALSA: pcm: remove SNDRV_PCM_IOCTL1_INFO internal command"), there was a possibility to pass information about the synchronized streams to the user space. The mentioned commit removed blindly the appropriate code with an irrelevant comment. The revert may be appropriate, but since this API was lost for several years without any complains, it's time to improve it. The hardware parameters may change the used stream clock source (e.g. USB hardware) so move this synchronization ID to hw_params as read-only field. It seems that pipewire can benefit from this API (disable adaptive resampling for perfectly synchronized PCM streams) now. Note that the contents of ID is not supposed to be used for direct comparison with a specific byte sequence. The "empty" case is when all bytes are zero (driver does not offer this information) and all other cases must be only used for equal comparison among PCM streams (including different sound cards) if they are using identical hardware clock. Cc: Takashi Sakamoto Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240625172836.589380-2-perex@perex.cz --- include/sound/pcm.h | 3 ++- include/uapi/sound/asound.h | 9 +++++---- 2 files changed, 7 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3edd7a7346da..dbce137d8806 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -93,6 +93,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2 /* 3 is absent slot. */ #define SNDRV_PCM_IOCTL1_FIFO_SIZE 4 +#define SNDRV_PCM_IOCTL1_SYNC_ID 5 #define SNDRV_PCM_TRIGGER_STOP 0 #define SNDRV_PCM_TRIGGER_START 1 @@ -401,7 +402,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ snd_pcm_uframes_t silence_filled; /* already filled part of silence area */ - union snd_pcm_sync_id sync; /* hardware synchronization ID */ + unsigned char sync[16]; /* hardware synchronization ID */ /* -- mmap -- */ struct snd_pcm_mmap_status *status; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 628d46a0da92..8bf7e8a0eb6f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -142,7 +142,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 17) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 18) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -334,7 +334,7 @@ union snd_pcm_sync_id { unsigned char id[16]; unsigned short id16[8]; unsigned int id32[4]; -}; +} __attribute__((deprecated)); struct snd_pcm_info { unsigned int device; /* RO/WR (control): device number */ @@ -348,7 +348,7 @@ struct snd_pcm_info { int dev_subclass; /* SNDRV_PCM_SUBCLASS_* */ unsigned int subdevices_count; unsigned int subdevices_avail; - union snd_pcm_sync_id sync; /* hardware synchronization ID */ + unsigned char pad1[16]; /* was: hardware synchronization ID */ unsigned char reserved[64]; /* reserved for future... */ }; @@ -420,7 +420,8 @@ 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 { -- cgit v1.2.3 From d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 25 Jun 2024 19:25:47 +0200 Subject: ALSA: pcm: optimize and clarify stream synchronization ID API Optimize the memory usage in struct snd_pcm_runtime - use boolean value for the standard sync ID scheme. Introduce snd_pcm_set_sync_per_card function to build synchronization IDs. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240625172836.589380-3-perex@perex.cz --- include/sound/pcm.h | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dbce137d8806..ac8f3aef9205 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,7 +402,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t silence_start; /* starting pointer to silence area */ snd_pcm_uframes_t silence_filled; /* already filled part of silence area */ - unsigned char sync[16]; /* hardware synchronization ID */ + bool std_sync_id; /* hardware synchronization - standard per card ID */ /* -- mmap -- */ struct snd_pcm_mmap_status *status; @@ -1156,7 +1156,18 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, const struct snd_pcm_ops *ops); -void snd_pcm_set_sync(struct snd_pcm_substream *substream); +void snd_pcm_set_sync_per_card(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, + const unsigned char *id, unsigned int len); +/** + * snd_pcm_set_sync - set the PCM sync id + * @substream: the pcm substream + * + * Use the default PCM sync identifier for the specific card. + */ +static inline void snd_pcm_set_sync(struct snd_pcm_substream *substream) +{ + substream->runtime->std_sync_id = true; +} int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); void snd_pcm_period_elapsed_under_stream_lock(struct snd_pcm_substream *substream); -- cgit v1.2.3 From 244389bd42870640c4b5ef672a360da329b579ed Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 3 Jul 2024 10:55:17 +0100 Subject: ASoC: cs35l56: Limit Speaker Volume to +12dB maximum Change CS35L56_MAIN_RENDER_USER_VOLUME_MAX to 48, to limit the maximum value of the Speaker Volume control to +12dB. The minimum value is unchanged so that the default 0dB has the same integer control value. The original maximum of 400 (+100dB) was the largest value that can be mathematically handled by the DSP. The actual maximum amplification is +12dB. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20240703095517.208077-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 642ef690ebc2..a6aa112e5741 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -207,7 +207,7 @@ /* CS35L56_MAIN_RENDER_USER_VOLUME */ #define CS35L56_MAIN_RENDER_USER_VOLUME_MIN -400 -#define CS35L56_MAIN_RENDER_USER_VOLUME_MAX 400 +#define CS35L56_MAIN_RENDER_USER_VOLUME_MAX 48 #define CS35L56_MAIN_RENDER_USER_VOLUME_MASK 0x0000FFC0 #define CS35L56_MAIN_RENDER_USER_VOLUME_SHIFT 6 #define CS35L56_MAIN_RENDER_USER_VOLUME_SIGNBIT 9 -- cgit v1.2.3 From bf95919fe1917efa8f5da83057ff9fc11130aa55 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 1 Jul 2024 09:39:36 +0200 Subject: ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() Number of widgets in array passed to snd_soc_dapm_new_controls() cannot be negative, so make it explicit by using 'unsigned int', just like snd_soc_add_component_controls() is doing. Reviewed-by: Dmitry Baryshkov Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240701-b4-qcom-audio-lpass-codec-cleanups-v3-4-6d98d4dd1ef5@linaro.org Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 667ecd4daa68..12cd7b5a2202 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -457,7 +457,7 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *uncontrol); int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget, int num); + const struct snd_soc_dapm_widget *widget, unsigned int num); struct snd_soc_dapm_widget *snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); struct snd_soc_dapm_widget *snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, -- cgit v1.2.3 From 37797c605da33445adc112561695f70bfaa11133 Mon Sep 17 00:00:00 2001 From: Herve Codina Date: Mon, 1 Jul 2024 13:30:34 +0200 Subject: soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle list qmc_chan_get_byphandle() and the resource managed version retrieve a channel from a simple phandle. Extend the API and introduce qmc_chan_get_byphandles_index() and the resource managed version in order to retrieve a channel from a phandle list using the provided index to identify the phandle in the list. Also update qmc_chan_get_byphandle() and the resource managed version to use qmc_chan_get_byphandles_index() and so avoid code duplication. Signed-off-by: Herve Codina Link: https://patch.msgid.link/20240701113038.55144-8-herve.codina@bootlin.com Signed-off-by: Mark Brown --- include/soc/fsl/qe/qmc.h | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/soc/fsl/qe/qmc.h b/include/soc/fsl/qe/qmc.h index 2a333fc1ea81..0fa7205145ce 100644 --- a/include/soc/fsl/qe/qmc.h +++ b/include/soc/fsl/qe/qmc.h @@ -16,11 +16,30 @@ struct device_node; struct device; struct qmc_chan; -struct qmc_chan *qmc_chan_get_byphandle(struct device_node *np, const char *phandle_name); +struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np, + const char *phandles_name, + int index); +struct qmc_chan *devm_qmc_chan_get_byphandles_index(struct device *dev, + struct device_node *np, + const char *phandles_name, + int index); + +static inline struct qmc_chan *qmc_chan_get_byphandle(struct device_node *np, + const char *phandle_name) +{ + return qmc_chan_get_byphandles_index(np, phandle_name, 0); +} + +static inline struct qmc_chan *devm_qmc_chan_get_byphandle(struct device *dev, + struct device_node *np, + const char *phandle_name) +{ + return devm_qmc_chan_get_byphandles_index(dev, np, phandle_name, 0); +} + struct qmc_chan *qmc_chan_get_bychild(struct device_node *np); void qmc_chan_put(struct qmc_chan *chan); -struct qmc_chan *devm_qmc_chan_get_byphandle(struct device *dev, struct device_node *np, - const char *phandle_name); + struct qmc_chan *devm_qmc_chan_get_bychild(struct device *dev, struct device_node *np); enum qmc_mode { -- cgit v1.2.3 From af8432b2e41abc0a20bdc01a3b144ea7b2f1ee09 Mon Sep 17 00:00:00 2001 From: Herve Codina Date: Mon, 1 Jul 2024 13:30:35 +0200 Subject: soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles() No function in the QMC API is available to get the number of phandles present in a phandle list. Fill this lack introducing qmc_chan_count_phandles(). Signed-off-by: Herve Codina Link: https://patch.msgid.link/20240701113038.55144-9-herve.codina@bootlin.com Signed-off-by: Mark Brown --- include/soc/fsl/qe/qmc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/soc/fsl/qe/qmc.h b/include/soc/fsl/qe/qmc.h index 0fa7205145ce..294e42ea8d4c 100644 --- a/include/soc/fsl/qe/qmc.h +++ b/include/soc/fsl/qe/qmc.h @@ -16,6 +16,8 @@ struct device_node; struct device; struct qmc_chan; +int qmc_chan_count_phandles(struct device_node *np, const char *phandles_name); + struct qmc_chan *qmc_chan_get_byphandles_index(struct device_node *np, const char *phandles_name, int index); -- cgit v1.2.3 From fefbbdfb59d3a20fd98734363e6dd9fa7cc65c70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jul 2024 18:03:42 +0200 Subject: ALSA: seq: Add tempo base unit for MIDI2 Set Tempo messages MIDI2 Set Tempo message defines the tempo in 10ns unit for finer accuracy, while MIDI1 was defined in 1us unit. For adapting this different unit, introduce "tempo_base" field to snd_seq_queue_tempo struct so that user-space can pass the proper tempo base unit. The accepted value is limited, it must be either 0, 10 or 1000. The protocol version is bumped to 1.0.4 along with this. The access with the older protocol version ignores the tempo-base value in ioctls and always treats as 1000. Reviewed-by: Jaroslav Kysela Link: https://patch.msgid.link/20240705160344.6481-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/uapi/sound/asequencer.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index c85fdd8895d8..39b37edcf813 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -10,7 +10,7 @@ #include /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 3) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 4) /** * definition of sequencer event types @@ -523,11 +523,12 @@ struct snd_seq_queue_status { /* queue tempo */ struct snd_seq_queue_tempo { int queue; /* sequencer queue */ - unsigned int tempo; /* current tempo, us/tick */ + unsigned int tempo; /* current tempo, us/tick (or different time-base below) */ int ppq; /* time resolution, ticks/quarter */ unsigned int skew_value; /* queue skew */ unsigned int skew_base; /* queue skew base */ - char reserved[24]; /* for the future */ + unsigned short tempo_base; /* tempo base in nsec unit; either 10 or 1000 */ + char reserved[22]; /* for the future */ }; -- cgit v1.2.3 From be5db7581f59621ed9cb9cbf6bebccda38263eb5 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Tue, 9 Jul 2024 12:33:40 +0800 Subject: ASoc: TAS2781: rename the tas2781_reset as tasdevice_reset Rename the tas2781_reset as tasdevice_reset in case of misunderstanding. RESET register for both tas2563 and tas2781 is same and the use of reset pin is also same. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240709043342.946-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index cd8ce522b78e..a43ad6dcb7c7 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 @@ -43,8 +43,8 @@ (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(0x0, 0X0, 0x01) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) @@ -140,7 +140,7 @@ struct tasdevice_priv { void (*apply_calibration)(struct tasdevice_priv *tas_priv); }; -void tas2781_reset(struct tasdevice_priv *tas_dev); +void tasdevice_reset(struct tasdevice_priv *tas_dev); int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, struct module *module, void (*cont)(const struct firmware *fw, void *context)); -- cgit v1.2.3 From 584e86e14c59d36688633002613792923620d8c0 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 10 Jul 2024 11:36:38 +0100 Subject: firmware: cs_dsp: Make wmfw and bin filename arguments const char * The wmfw_filename and bin_filename strings passed into cs_dsp_power_up() and cs_dsp_adsp1_power_up() should be const char *. Signed-off-by: Richard Fitzgerald Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20240710103640.78197-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 82687e07a7c2..8078dc377948 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -213,13 +213,13 @@ int cs_dsp_adsp2_init(struct cs_dsp *dsp); int cs_dsp_halo_init(struct cs_dsp *dsp); int cs_dsp_adsp1_power_up(struct cs_dsp *dsp, - const struct firmware *wmfw_firmware, char *wmfw_filename, - const struct firmware *coeff_firmware, char *coeff_filename, + const struct firmware *wmfw_firmware, const char *wmfw_filename, + const struct firmware *coeff_firmware, const char *coeff_filename, const char *fw_name); void cs_dsp_adsp1_power_down(struct cs_dsp *dsp); int cs_dsp_power_up(struct cs_dsp *dsp, - const struct firmware *wmfw_firmware, char *wmfw_filename, - const struct firmware *coeff_firmware, char *coeff_filename, + const struct firmware *wmfw_firmware, const char *wmfw_filename, + const struct firmware *coeff_firmware, const char *coeff_filename, const char *fw_name); void cs_dsp_power_down(struct cs_dsp *dsp); int cs_dsp_run(struct cs_dsp *dsp); -- cgit v1.2.3 From dc0e5ca8856dc6a97e3b117879dfb2b52bda06b6 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 10 Jul 2024 11:36:40 +0100 Subject: firmware: cs_dsp: Rename fw_ver to wmfw_ver Rename the confusingly named struct member fw_ver to wmfw_ver. It contains the wmfw format version of the loaded wmfw file. This commit also contains an update to wm_adsp for the new name. Signed-off-by: Richard Fitzgerald Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20240710103640.78197-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/linux/firmware/cirrus/cs_dsp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 8078dc377948..2e649810169f 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -167,7 +167,7 @@ struct cs_dsp { const struct cs_dsp_region *mem; int num_mems; - int fw_ver; + int wmfw_ver; bool booted; bool running; -- cgit v1.2.3 From 75ed63a5ab5d1d2872c735bc7edf8fef0e2fa2ea Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Wed, 10 Jul 2024 14:42:37 +0800 Subject: ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume Requriment from customer to add new kcontrol to set tas2563 digital Volume Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240710064238.1480-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781-tlv.h | 262 +++++++++++++++++++++++++++++++++++++++++++- include/sound/tas2781.h | 1 + 2 files changed, 262 insertions(+), 1 deletion(-) (limited to 'include') 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 a43ad6dcb7c7..18161d02a96f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -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) -- cgit v1.2.3