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authorLinus Torvalds <torvalds@linux-foundation.org>2024-01-12 22:55:28 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2024-01-12 22:55:28 +0300
commit7912a6391f3ee7eb9f9a69227a209d502679bc0c (patch)
tree38be70147934f98cde0ccef0ba65b836246e4d0d /include
parentcf65598d5909acf5e7b7dc9e21786e386356bc81 (diff)
parentfd38dd6abda589a8771e7872e4dea28c99c6a6ef (diff)
downloadlinux-7912a6391f3ee7eb9f9a69227a209d502679bc0c.tar.xz
Merge tag 'sound-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "It was a calm development cycle. There were an ALSA core extension for subformat PCM bits and a few ASoC core changes to support N:M mappings, while the most of remaining changes are driver-specific. Core: - API extensions for properly limiting PCM format bits via subformat - Enhanced support for N:M CPU:CODEC mappings in the core and in audio-graph-card2 ASoC: - Lots of SOF updates: fallback support to older IPC versions, notification on control changes with IPC4. Also supports for ACPI parse for the ES83xx driver that reduces quirks. - Device tree support for describing parts of the card which can be active over suspend (for very low power playback or wake word use cases) - Support for more AMD and Intel systems, NXP i.MX8m MICFIL, Qualcomm SM8250, SM8550, SM8650 and X1E80100 - Drop of Freescale MPC8610 code that is no longer supported HD-audio: - More CS35L41 codec extensions for Dell, HP and Lenovo models - TAS2781 codec extensions for Lenovo and co - New PCM subformat supports Others: - More enhancement for Scarlett2 USB mixer support - Various kselftest fixes" * tag 'sound-6.8-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (337 commits) kselftest/alsa - conf: Stringify the printed errno in sysfs_get() kselftest/alsa - mixer-test: Fix the print format specifier warning kselftest/alsa - mixer-test: Fix the print format specifier warning kselftest/alsa - mixer-test: fix the number of parameters to ksft_exit_fail_msg() ALSA: hda/tas2781: annotate calibration data endianness ALSA: hda/realtek: Fix mute and mic-mute LEDs for HP Envy X360 13-ay0xxx ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140 ALSA: ac97: fix build regression ALSA: hda: cs35l41: Support more HP models without _DSD ALSA: hda/tas2781: add fixup for Lenovo 14ARB7 ALSA: hda/tas2781: add TAS2563 support for 14ARB7 ALSA: hda/tas2781: add configurable global i2c address ALSA: hda/tas2781: add ptrs to calibration functions ALSA: hda: Add driver properties for cs35l41 for Lenovo Legion Slim 7 Gen 8 serie ALSA: hda/realtek: enable SND_PCI_QUIRK for Lenovo Legion Slim 7 Gen 8 (2023) serie ALSA: hda/tas2781: configure the amp after firmware load ALSA: mark all struct bus_type as const ASoC: pxa: sspa: Don't select SND_ARM ASoC: rt5663: cancel the work when system suspends ALSA: scarlett2: Add PCM Input Switch for Solo Gen 4 ...
Diffstat (limited to 'include')
-rw-r--r--include/linux/pci_ids.h1
-rw-r--r--include/sound/ac97_codec.h2
-rw-r--r--include/sound/cs4271.h1
-rw-r--r--include/sound/hda_codec.h5
-rw-r--r--include/sound/hdaudio.h15
-rw-r--r--include/sound/pcm.h7
-rw-r--r--include/sound/pcm_params.h2
-rw-r--r--include/sound/rt5682s.h8
-rw-r--r--include/sound/simple_card_utils.h3
-rw-r--r--include/sound/soc.h59
-rw-r--r--include/sound/sof.h15
-rw-r--r--include/sound/sof/dai-imx.h7
-rw-r--r--include/sound/sof/dai.h2
-rw-r--r--include/sound/sof/ipc4/header.h35
-rw-r--r--include/sound/sof/topology.h61
-rw-r--r--include/sound/tas2781.h8
-rw-r--r--include/sound/wm0010.h6
-rw-r--r--include/sound/wm1250-ev1.h24
-rw-r--r--include/sound/wm2200.h2
-rw-r--r--include/sound/wm5100.h4
-rw-r--r--include/sound/wm8996.h3
-rw-r--r--include/uapi/sound/asound.h7
-rw-r--r--include/uapi/sound/scarlett2.h54
-rw-r--r--include/uapi/sound/sof/tokens.h5
24 files changed, 253 insertions, 83 deletions
diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h
index 844ffdac8d7d..a0c75e467df3 100644
--- a/include/linux/pci_ids.h
+++ b/include/linux/pci_ids.h
@@ -3067,6 +3067,7 @@
#define PCI_DEVICE_ID_INTEL_82443GX_0 0x71a0
#define PCI_DEVICE_ID_INTEL_82443GX_2 0x71a2
#define PCI_DEVICE_ID_INTEL_82372FB_1 0x7601
+#define PCI_DEVICE_ID_INTEL_HDA_ARL 0x7728
#define PCI_DEVICE_ID_INTEL_HDA_RPL_S 0x7a50
#define PCI_DEVICE_ID_INTEL_HDA_ADL_S 0x7ad0
#define PCI_DEVICE_ID_INTEL_HDA_MTL 0x7e28
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index c495c6d5fbe0..4bd3be3a3192 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -410,7 +410,7 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm);
int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
/* ad hoc AC97 device driver access */
-extern struct bus_type ac97_bus_type;
+extern const struct bus_type ac97_bus_type;
/* AC97 platform_data adding function */
static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h
index 6ff23ab48894..5a55d135bdea 100644
--- a/include/sound/cs4271.h
+++ b/include/sound/cs4271.h
@@ -9,7 +9,6 @@
#define __CS4271_H
struct cs4271_platform_data {
- int gpio_nreset; /* GPIO driving Reset pin, if any */
bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */
/*
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index 5497dc9c396a..9c94ba7c183d 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -141,6 +141,7 @@ struct hda_pcm_stream {
hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
u32 rates; /* supported rates */
u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */
+ u32 subformats; /* for S32_LE format, SNDRV_PCM_SUBFMTBIT_* */
unsigned int maxbps; /* supported max. bit per sample */
const struct snd_pcm_chmap_elem *chmap; /* chmap to override */
struct hda_pcm_ops ops;
@@ -448,8 +449,8 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
#define snd_hda_codec_cleanup_stream(codec, nid) \
__snd_hda_codec_cleanup_stream(codec, nid, 0)
-#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, bpsp) \
- snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, bpsp)
+#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, subfmtp, bpsp) \
+ snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, subfmtp, bpsp)
#define snd_hda_is_supported_format(codec, nid, fmt) \
snd_hdac_is_supported_format(&(codec)->core, nid, fmt)
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index dd7c87bbc613..a73d7f34f4e5 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -33,7 +33,7 @@ struct hda_device_id;
/*
* exported bus type
*/
-extern struct bus_type snd_hda_bus_type;
+extern const struct bus_type snd_hda_bus_type;
/*
* generic arrays
@@ -140,13 +140,14 @@ int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid,
hda_nid_t *start_id);
-unsigned int snd_hdac_calc_stream_format(unsigned int rate,
- unsigned int channels,
- snd_pcm_format_t format,
- unsigned int maxbps,
- unsigned short spdif_ctls);
+unsigned int snd_hdac_stream_format_bits(snd_pcm_format_t format, snd_pcm_subformat_t subformat,
+ unsigned int maxbits);
+unsigned int snd_hdac_stream_format(unsigned int channels, unsigned int bits, unsigned int rate);
+unsigned int snd_hdac_spdif_stream_format(unsigned int channels, unsigned int bits,
+ unsigned int rate, unsigned short spdif_ctls);
int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid,
- u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
+ u32 *ratesp, u64 *formatsp, u32 *subformatsp,
+ unsigned int *bpsp);
bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
unsigned int format);
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 2a815373dac1..cc175c623dac 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -32,6 +32,7 @@
struct snd_pcm_hardware {
unsigned int info; /* SNDRV_PCM_INFO_* */
u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ u32 subformats; /* for S32_LE, SNDRV_PCM_SUBFMTBIT_* */
unsigned int rates; /* SNDRV_PCM_RATE_* */
unsigned int rate_min; /* min rate */
unsigned int rate_max; /* max rate */
@@ -217,6 +218,12 @@ struct snd_pcm_ops {
#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE
#endif
+#define _SNDRV_PCM_SUBFMTBIT(fmt) BIT((__force int)SNDRV_PCM_SUBFORMAT_##fmt)
+#define SNDRV_PCM_SUBFMTBIT_STD _SNDRV_PCM_SUBFMTBIT(STD)
+#define SNDRV_PCM_SUBFMTBIT_MSBITS_MAX _SNDRV_PCM_SUBFMTBIT(MSBITS_MAX)
+#define SNDRV_PCM_SUBFMTBIT_MSBITS_20 _SNDRV_PCM_SUBFMTBIT(MSBITS_20)
+#define SNDRV_PCM_SUBFMTBIT_MSBITS_24 _SNDRV_PCM_SUBFMTBIT(MSBITS_24)
+
struct snd_pcm_file {
struct snd_pcm_substream *substream;
int no_compat_mmap;
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index ba184f49f7e1..fbf35df6e5cf 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -362,6 +362,8 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p)
return snd_pcm_format_physical_width(params_format(p));
}
+int snd_pcm_hw_params_bits(const struct snd_pcm_hw_params *p);
+
static inline void
params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt)
{
diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h
index 66ca0c75b914..006e6003d11c 100644
--- a/include/sound/rt5682s.h
+++ b/include/sound/rt5682s.h
@@ -31,6 +31,13 @@ enum rt5682s_dai_clks {
RT5682S_DAI_NUM_CLKS,
};
+enum {
+ RT5682S_LDO_1_607V,
+ RT5682S_LDO_1_5V,
+ RT5682S_LDO_1_406V,
+ RT5682S_LDO_1_731V,
+};
+
struct rt5682s_platform_data {
enum rt5682s_dmic1_data_pin dmic1_data_pin;
enum rt5682s_dmic1_clk_pin dmic1_clk_pin;
@@ -38,6 +45,7 @@ struct rt5682s_platform_data {
unsigned int dmic_clk_rate;
unsigned int dmic_delay;
unsigned int amic_delay;
+ unsigned int ldo_dacref;
bool dmic_clk_driving_high;
const char *dai_clk_names[RT5682S_DAI_NUM_CLKS];
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index e5da10b4c43b..ad67957b7b48 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -195,6 +195,9 @@ 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,
+ bool *is_playback_only, bool *is_capture_only);
+
#ifdef DEBUG
static inline void simple_util_debug_dai(struct simple_util_priv *priv,
char *name,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 7792c393e238..6defc5547ff9 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -620,6 +620,7 @@ enum snd_soc_trigger_order {
struct snd_soc_pcm_stream {
const char *stream_name;
u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ u32 subformats; /* for S32_LE format, SNDRV_PCM_SUBFMTBIT_* */
unsigned int rates; /* SNDRV_PCM_RATE_* */
unsigned int rate_min; /* min rate */
unsigned int rate_max; /* max rate */
@@ -655,8 +656,45 @@ struct snd_soc_dai_link_component {
struct of_phandle_args *dai_args;
};
-struct snd_soc_dai_link_codec_ch_map {
- unsigned int connected_cpu_id;
+/*
+ * [dai_link->ch_maps Image sample]
+ *
+ *-------------------------
+ * CPU0 <---> Codec0
+ *
+ * ch-map[0].cpu = 0 ch-map[0].codec = 0
+ *
+ *-------------------------
+ * CPU0 <---> Codec0
+ * CPU1 <---> Codec1
+ * CPU2 <---> Codec2
+ *
+ * ch-map[0].cpu = 0 ch-map[0].codec = 0
+ * ch-map[1].cpu = 1 ch-map[1].codec = 1
+ * ch-map[2].cpu = 2 ch-map[2].codec = 2
+ *
+ *-------------------------
+ * CPU0 <---> Codec0
+ * CPU1 <-+-> Codec1
+ * CPU2 <-/
+ *
+ * ch-map[0].cpu = 0 ch-map[0].codec = 0
+ * ch-map[1].cpu = 1 ch-map[1].codec = 1
+ * ch-map[2].cpu = 2 ch-map[2].codec = 1
+ *
+ *-------------------------
+ * CPU0 <---> Codec0
+ * CPU1 <-+-> Codec1
+ * \-> Codec2
+ *
+ * ch-map[0].cpu = 0 ch-map[0].codec = 0
+ * ch-map[1].cpu = 1 ch-map[1].codec = 1
+ * ch-map[2].cpu = 1 ch-map[2].codec = 2
+ *
+ */
+struct snd_soc_dai_link_ch_map {
+ unsigned int cpu;
+ unsigned int codec;
unsigned int ch_mask;
};
@@ -688,7 +726,9 @@ struct snd_soc_dai_link {
struct snd_soc_dai_link_component *codecs;
unsigned int num_codecs;
- struct snd_soc_dai_link_codec_ch_map *codec_ch_maps;
+ /* num_ch_maps = max(num_cpu, num_codecs) */
+ struct snd_soc_dai_link_ch_map *ch_maps;
+
/*
* You MAY specify the link's platform/PCM/DMA driver, either by
* device name, or by DT/OF node, but not both. Some forms of link
@@ -775,6 +815,10 @@ struct snd_soc_dai_link {
#endif
};
+static inline int snd_soc_link_num_ch_map(struct snd_soc_dai_link *link) {
+ return max(link->num_cpus, link->num_codecs);
+}
+
static inline struct snd_soc_dai_link_component*
snd_soc_link_to_cpu(struct snd_soc_dai_link *link, int n) {
return &(link)->cpus[n];
@@ -808,6 +852,12 @@ snd_soc_link_to_platform(struct snd_soc_dai_link *link, int n) {
((cpu) = snd_soc_link_to_cpu(link, i)); \
(i)++)
+#define for_each_link_ch_maps(link, i, ch_map) \
+ for ((i) = 0; \
+ ((i) < snd_soc_link_num_ch_map(link) && \
+ ((ch_map) = link->ch_maps + i)); \
+ (i)++)
+
/*
* Sample 1 : Single CPU/Codec/Platform
*
@@ -889,7 +939,7 @@ snd_soc_link_to_platform(struct snd_soc_dai_link *link, int n) {
#define COMP_PLATFORM(_name) { .name = _name }
#define COMP_AUX(_name) { .name = _name }
#define COMP_CODEC_CONF(_name) { .name = _name }
-#define COMP_DUMMY() { .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", }
+#define COMP_DUMMY() /* see snd_soc_fill_dummy_dai() */
extern struct snd_soc_dai_link_component null_dailink_component[0];
extern struct snd_soc_dai_link_component snd_soc_dummy_dlc;
@@ -1163,6 +1213,7 @@ struct snd_soc_pcm_runtime {
((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) && \
((dai) = (rtd)->dais[i]); \
(i)++)
+#define for_each_rtd_ch_maps(rtd, i, ch_maps) for_each_link_ch_maps(rtd->dai_link, i, ch_maps)
void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd);
diff --git a/include/sound/sof.h b/include/sound/sof.h
index 268d0ca0f69f..05213bb515a3 100644
--- a/include/sound/sof.h
+++ b/include/sound/sof.h
@@ -57,6 +57,18 @@ enum sof_ipc_type {
SOF_IPC_TYPE_COUNT
};
+struct sof_loadable_file_profile {
+ enum sof_ipc_type ipc_type;
+
+ const char *fw_path;
+ const char *fw_path_postfix;
+ const char *fw_name;
+ const char *fw_lib_path;
+ const char *fw_lib_path_postfix;
+ const char *tplg_path;
+ const char *tplg_name;
+};
+
/*
* SOF Platform data.
*/
@@ -86,6 +98,9 @@ struct snd_sof_pdata {
/* descriptor */
const struct sof_dev_desc *desc;
+ /* platform's preferred IPC type and path overrides */
+ struct sof_loadable_file_profile ipc_file_profile_base;
+
/* firmware and topology filenames */
const char *fw_filename_prefix;
const char *fw_filename;
diff --git a/include/sound/sof/dai-imx.h b/include/sound/sof/dai-imx.h
index ca8325353d41..6bc987bd4761 100644
--- a/include/sound/sof/dai-imx.h
+++ b/include/sound/sof/dai-imx.h
@@ -51,4 +51,11 @@ struct sof_ipc_dai_sai_params {
uint16_t tdm_slot_width;
uint16_t reserved2; /* alignment */
} __packed;
+
+/* MICFIL Configuration Request - SOF_IPC_DAI_MICFIL_CONFIG */
+struct sof_ipc_dai_micfil_params {
+ uint32_t pdm_rate;
+ uint32_t pdm_ch;
+} __packed;
+
#endif
diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index 3041f5805b7b..4773a5f616a4 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -88,6 +88,7 @@ enum sof_ipc_dai_type {
SOF_DAI_AMD_HS, /**< Amd HS */
SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */
SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */
+ SOF_DAI_IMX_MICFIL, /** < i.MX MICFIL PDM */
};
/* general purpose DAI configuration */
@@ -117,6 +118,7 @@ struct sof_ipc_dai_config {
struct sof_ipc_dai_acpdmic_params acpdmic;
struct sof_ipc_dai_acp_params acphs;
struct sof_ipc_dai_mtk_afe_params afe;
+ struct sof_ipc_dai_micfil_params micfil;
};
} __packed;
diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h
index 574a9d581f88..1eb538e18236 100644
--- a/include/sound/sof/ipc4/header.h
+++ b/include/sound/sof/ipc4/header.h
@@ -423,6 +423,12 @@ enum sof_ipc4_fw_config_params {
SOF_IPC4_FW_CFG_RESERVED,
SOF_IPC4_FW_CFG_POWER_GATING_POLICY,
SOF_IPC4_FW_CFG_ASSERT_MODE,
+ SOF_IPC4_FW_RESERVED1,
+ SOF_IPC4_FW_RESERVED2,
+ SOF_IPC4_FW_RESERVED3,
+ SOF_IPC4_FW_RESERVED4,
+ SOF_IPC4_FW_RESERVED5,
+ SOF_IPC4_FW_CONTEXT_SAVE
};
struct sof_ipc4_fw_version {
@@ -532,6 +538,35 @@ struct sof_ipc4_notify_resource_data {
#define SOF_IPC4_DEBUG_SLOT_TELEMETRY 0x4c455400
#define SOF_IPC4_DEBUG_SLOT_BROKEN 0x44414544
+/**
+ * struct sof_ipc4_notify_module_data - payload for module notification
+ * @instance_id: instance ID of the originator module of the notification
+ * @module_id: module ID of the originator of the notification
+ * @event_id: module specific event id
+ * @event_data_size: Size of the @event_data (if any) in bytes
+ * @event_data: Optional notification data, module and notification dependent
+ */
+struct sof_ipc4_notify_module_data {
+ uint16_t instance_id;
+ uint16_t module_id;
+ uint32_t event_id;
+ uint32_t event_data_size;
+ uint8_t event_data[];
+} __packed __aligned(4);
+
+/*
+ * ALSA kcontrol change notification
+ *
+ * The event_id of struct sof_ipc4_notify_module_data is divided into two u16:
+ * upper u16: magic number for ALSA kcontrol types: 0xA15A
+ * lower u16: param_id of the control, which is the type of the control
+ * The event_data contains the struct sof_ipc4_control_msg_payload of the control
+ * which sent the notification.
+ */
+#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_MASK GENMASK(31, 16)
+#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_VAL 0xA15A0000
+#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_PARAMID_MASK GENMASK(15, 0)
+
/** @}*/
#endif
diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h
index 906e2f327ad2..b3ca886fa28f 100644
--- a/include/sound/sof/topology.h
+++ b/include/sound/sof/topology.h
@@ -39,6 +39,7 @@ enum sof_comp_type {
SOF_COMP_ASRC, /**< Asynchronous sample rate converter */
SOF_COMP_DCBLOCK,
SOF_COMP_SMART_AMP, /**< smart amplifier component */
+ SOF_COMP_MODULE_ADAPTER, /**< module adapter */
/* keep FILEREAD/FILEWRITE as the last ones */
SOF_COMP_FILEREAD = 10000, /**< host test based file IO */
SOF_COMP_FILEWRITE = 10001, /**< host test based file IO */
@@ -59,7 +60,7 @@ struct sof_ipc_comp {
/* extended data length, 0 if no extended data */
uint32_t ext_data_length;
-} __packed;
+} __packed __aligned(4);
/*
* Component Buffers
@@ -68,14 +69,15 @@ struct sof_ipc_comp {
/*
* SOF memory capabilities, add new ones at the end
*/
-#define SOF_MEM_CAPS_RAM (1 << 0)
-#define SOF_MEM_CAPS_ROM (1 << 1)
-#define SOF_MEM_CAPS_EXT (1 << 2) /**< external */
-#define SOF_MEM_CAPS_LP (1 << 3) /**< low power */
-#define SOF_MEM_CAPS_HP (1 << 4) /**< high performance */
-#define SOF_MEM_CAPS_DMA (1 << 5) /**< DMA'able */
-#define SOF_MEM_CAPS_CACHE (1 << 6) /**< cacheable */
-#define SOF_MEM_CAPS_EXEC (1 << 7) /**< executable */
+#define SOF_MEM_CAPS_RAM BIT(0)
+#define SOF_MEM_CAPS_ROM BIT(1)
+#define SOF_MEM_CAPS_EXT BIT(2) /**< external */
+#define SOF_MEM_CAPS_LP BIT(3) /**< low power */
+#define SOF_MEM_CAPS_HP BIT(4) /**< high performance */
+#define SOF_MEM_CAPS_DMA BIT(5) /**< DMA'able */
+#define SOF_MEM_CAPS_CACHE BIT(6) /**< cacheable */
+#define SOF_MEM_CAPS_EXEC BIT(7) /**< executable */
+#define SOF_MEM_CAPS_L3 BIT(8) /**< L3 memory */
/*
* overrun will cause ring buffer overwrite, instead of XRUN.
@@ -87,6 +89,9 @@ struct sof_ipc_comp {
*/
#define SOF_BUF_UNDERRUN_PERMITTED BIT(1)
+/* the UUID size in bytes, shared between FW and host */
+#define SOF_UUID_SIZE 16
+
/* create new component buffer - SOF_IPC_TPLG_BUFFER_NEW */
struct sof_ipc_buffer {
struct sof_ipc_comp comp;
@@ -94,7 +99,7 @@ struct sof_ipc_buffer {
uint32_t caps; /**< SOF_MEM_CAPS_ */
uint32_t flags; /**< SOF_BUF_ flags defined above */
uint32_t reserved; /**< reserved for future use */
-} __packed;
+} __packed __aligned(4);
/* generic component config data - must always be after struct sof_ipc_comp */
struct sof_ipc_comp_config {
@@ -107,7 +112,7 @@ struct sof_ipc_comp_config {
/* reserved for future use */
uint32_t reserved[2];
-} __packed;
+} __packed __aligned(4);
/* generic host component */
struct sof_ipc_comp_host {
@@ -116,7 +121,7 @@ struct sof_ipc_comp_host {
uint32_t direction; /**< SOF_IPC_STREAM_ */
uint32_t no_irq; /**< don't send periodic IRQ to host/DSP */
uint32_t dmac_config; /**< DMA engine specific */
-} __packed;
+} __packed __aligned(4);
/* generic DAI component */
struct sof_ipc_comp_dai {
@@ -126,13 +131,13 @@ struct sof_ipc_comp_dai {
uint32_t dai_index; /**< index of this type dai */
uint32_t type; /**< DAI type - SOF_DAI_ */
uint32_t reserved; /**< reserved */
-} __packed;
+} __packed __aligned(4);
/* generic mixer component */
struct sof_ipc_comp_mixer {
struct sof_ipc_comp comp;
struct sof_ipc_comp_config config;
-} __packed;
+} __packed __aligned(4);
/* volume ramping types */
enum sof_volume_ramp {
@@ -140,6 +145,8 @@ enum sof_volume_ramp {
SOF_VOLUME_LOG,
SOF_VOLUME_LINEAR_ZC,
SOF_VOLUME_LOG_ZC,
+ SOF_VOLUME_WINDOWS_FADE,
+ SOF_VOLUME_WINDOWS_NO_FADE,
};
/* generic volume component */
@@ -151,7 +158,7 @@ struct sof_ipc_comp_volume {
uint32_t max_value;
uint32_t ramp; /**< SOF_VOLUME_ */
uint32_t initial_ramp; /**< ramp space in ms */
-} __packed;
+} __packed __aligned(4);
/* generic SRC component */
struct sof_ipc_comp_src {
@@ -161,7 +168,7 @@ struct sof_ipc_comp_src {
uint32_t source_rate; /**< source rate or 0 for variable */
uint32_t sink_rate; /**< sink rate or 0 for variable */
uint32_t rate_mask; /**< SOF_RATE_ supported rates */
-} __packed;
+} __packed __aligned(4);
/* generic ASRC component */
struct sof_ipc_comp_asrc {
@@ -187,13 +194,13 @@ struct sof_ipc_comp_asrc {
/* reserved for future use */
uint32_t reserved[4];
-} __attribute__((packed));
+} __packed __aligned(4);
/* generic MUX component */
struct sof_ipc_comp_mux {
struct sof_ipc_comp comp;
struct sof_ipc_comp_config config;
-} __packed;
+} __packed __aligned(4);
/* generic tone generator component */
struct sof_ipc_comp_tone {
@@ -208,7 +215,7 @@ struct sof_ipc_comp_tone {
int32_t period;
int32_t repeats;
int32_t ramp_step;
-} __packed;
+} __packed __aligned(4);
/** \brief Types of processing components */
enum sof_ipc_process_type {
@@ -234,8 +241,8 @@ struct sof_ipc_comp_process {
/* reserved for future use */
uint32_t reserved[7];
- uint8_t data[];
-} __packed;
+ unsigned char data[];
+} __packed __aligned(4);
/* frees components, buffers and pipelines
* SOF_IPC_TPLG_COMP_FREE, SOF_IPC_TPLG_PIPE_FREE, SOF_IPC_TPLG_BUFFER_FREE
@@ -243,13 +250,13 @@ struct sof_ipc_comp_process {
struct sof_ipc_free {
struct sof_ipc_cmd_hdr hdr;
uint32_t id;
-} __packed;
+} __packed __aligned(4);
struct sof_ipc_comp_reply {
struct sof_ipc_reply rhdr;
uint32_t id;
uint32_t offset;
-} __packed;
+} __packed __aligned(4);
/*
* Pipeline
@@ -274,25 +281,25 @@ struct sof_ipc_pipe_new {
uint32_t frames_per_sched;/**< output frames of pipeline, 0 is variable */
uint32_t xrun_limit_usecs; /**< report xruns greater than limit */
uint32_t time_domain; /**< scheduling time domain */
-} __packed;
+} __packed __aligned(4);
/* pipeline construction complete - SOF_IPC_TPLG_PIPE_COMPLETE */
struct sof_ipc_pipe_ready {
struct sof_ipc_cmd_hdr hdr;
uint32_t comp_id;
-} __packed;
+} __packed __aligned(4);
struct sof_ipc_pipe_free {
struct sof_ipc_cmd_hdr hdr;
uint32_t comp_id;
-} __packed;
+} __packed __aligned(4);
/* connect two components in pipeline - SOF_IPC_TPLG_COMP_CONNECT */
struct sof_ipc_pipe_comp_connect {
struct sof_ipc_cmd_hdr hdr;
uint32_t source_id;
uint32_t sink_id;
-} __packed;
+} __packed __aligned(4);
/* external events */
enum sof_event_types {
diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h
index a6c808b22318..0a86ab8d47b9 100644
--- a/include/sound/tas2781.h
+++ b/include/sound/tas2781.h
@@ -22,6 +22,7 @@
#define TAS2781_DRV_VER 1
#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)
@@ -121,6 +122,8 @@ struct tasdevice_priv {
bool force_fwload_status;
bool playback_started;
bool isacpi;
+ unsigned int global_addr;
+
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,
@@ -131,6 +134,9 @@ struct tasdevice_priv {
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);
};
void tas2781_reset(struct tasdevice_priv *tas_dev);
@@ -139,6 +145,8 @@ int tascodec_init(struct tasdevice_priv *tas_priv, void *codec,
struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c);
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_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/include/sound/wm0010.h b/include/sound/wm0010.h
index 13b473935ca1..14ff9056c5d0 100644
--- a/include/sound/wm0010.h
+++ b/include/sound/wm0010.h
@@ -11,12 +11,6 @@
#define WM0010_PDATA_H
struct wm0010_pdata {
- int gpio_reset;
-
- /* Set if there is an inverter between the GPIO controlling
- * the reset signal and the device.
- */
- int reset_active_high;
int irq_flags;
};
diff --git a/include/sound/wm1250-ev1.h b/include/sound/wm1250-ev1.h
deleted file mode 100644
index d16614ebecb4..000000000000
--- a/include/sound/wm1250-ev1.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1
- *
- * Copyright 2011 Wolfson Microelectronics. PLC.
- */
-
-#ifndef __LINUX_SND_WM1250_EV1_H
-#define __LINUX_SND_WM1250_EV1_H
-
-#define WM1250_EV1_NUM_GPIOS 5
-
-#define WM1250_EV1_GPIO_CLK_ENA 0
-#define WM1250_EV1_GPIO_CLK_SEL0 1
-#define WM1250_EV1_GPIO_CLK_SEL1 2
-#define WM1250_EV1_GPIO_OSR 3
-#define WM1250_EV1_GPIO_MASTER 4
-
-
-struct wm1250_ev1_pdata {
- int gpios[WM1250_EV1_NUM_GPIOS];
-};
-
-#endif
diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h
index 9987e6c09bdc..2e4913ee2505 100644
--- a/include/sound/wm2200.h
+++ b/include/sound/wm2200.h
@@ -42,8 +42,6 @@ struct wm2200_micbias {
};
struct wm2200_pdata {
- int reset; /** GPIO controlling /RESET, if any */
- int ldo_ena; /** GPIO controlling LODENA, if any */
int irq_flags;
int gpio_defaults[4];
diff --git a/include/sound/wm5100.h b/include/sound/wm5100.h
index b94badf72947..1c48090fdb2c 100644
--- a/include/sound/wm5100.h
+++ b/include/sound/wm5100.h
@@ -36,11 +36,7 @@ struct wm5100_jack_mode {
#define WM5100_GPIO_SET 0x10000
struct wm5100_pdata {
- int reset; /** GPIO controlling /RESET, if any */
- int ldo_ena; /** GPIO controlling LODENA, if any */
- int hp_pol; /** GPIO controlling headset polarity, if any */
int irq_flags;
- int gpio_base;
struct wm5100_jack_mode jack_modes[2];
diff --git a/include/sound/wm8996.h b/include/sound/wm8996.h
index 247f9917e33d..342abeef288f 100644
--- a/include/sound/wm8996.h
+++ b/include/sound/wm8996.h
@@ -33,8 +33,6 @@ struct wm8996_retune_mobile_config {
struct wm8996_pdata {
int irq_flags; /** Set IRQ trigger flags; default active low */
- int ldo_ena; /** GPIO for LDO1; -1 for none */
-
int micdet_def; /** Default MICDET_SRC/HP1FB_SRC/MICD_BIAS */
enum wm8996_inmode inl_mode;
@@ -42,7 +40,6 @@ struct wm8996_pdata {
u32 spkmute_seq; /** Value for register 0x802 */
- int gpio_base;
u32 gpio_default[5];
int num_retune_mobile_cfgs;
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index f9939da41122..d5b9cfbd9cea 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, 15)
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 16)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
@@ -267,7 +267,10 @@ typedef int __bitwise snd_pcm_format_t;
typedef int __bitwise snd_pcm_subformat_t;
#define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0)
-#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD
+#define SNDRV_PCM_SUBFORMAT_MSBITS_MAX ((__force snd_pcm_subformat_t) 1)
+#define SNDRV_PCM_SUBFORMAT_MSBITS_20 ((__force snd_pcm_subformat_t) 2)
+#define SNDRV_PCM_SUBFORMAT_MSBITS_24 ((__force snd_pcm_subformat_t) 3)
+#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_MSBITS_24
#define SNDRV_PCM_INFO_MMAP 0x00000001 /* hardware supports mmap */
#define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */
diff --git a/include/uapi/sound/scarlett2.h b/include/uapi/sound/scarlett2.h
new file mode 100644
index 000000000000..91467ab0ed70
--- /dev/null
+++ b/include/uapi/sound/scarlett2.h
@@ -0,0 +1,54 @@
+/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */
+/*
+ * Focusrite Scarlett 2 Protocol Driver for ALSA
+ * (including Scarlett 2nd Gen, 3rd Gen, 4th Gen, Clarett USB, and
+ * Clarett+ series products)
+ *
+ * Copyright (c) 2023 by Geoffrey D. Bennett <g at b4.vu>
+ */
+#ifndef __UAPI_SOUND_SCARLETT2_H
+#define __UAPI_SOUND_SCARLETT2_H
+
+#include <linux/types.h>
+#include <linux/ioctl.h>
+
+#define SCARLETT2_HWDEP_MAJOR 1
+#define SCARLETT2_HWDEP_MINOR 0
+#define SCARLETT2_HWDEP_SUBMINOR 0
+
+#define SCARLETT2_HWDEP_VERSION \
+ ((SCARLETT2_HWDEP_MAJOR << 16) | \
+ (SCARLETT2_HWDEP_MINOR << 8) | \
+ SCARLETT2_HWDEP_SUBMINOR)
+
+#define SCARLETT2_HWDEP_VERSION_MAJOR(v) (((v) >> 16) & 0xFF)
+#define SCARLETT2_HWDEP_VERSION_MINOR(v) (((v) >> 8) & 0xFF)
+#define SCARLETT2_HWDEP_VERSION_SUBMINOR(v) ((v) & 0xFF)
+
+/* Get protocol version */
+#define SCARLETT2_IOCTL_PVERSION _IOR('S', 0x60, int)
+
+/* Reboot */
+#define SCARLETT2_IOCTL_REBOOT _IO('S', 0x61)
+
+/* Select flash segment */
+#define SCARLETT2_SEGMENT_ID_SETTINGS 0
+#define SCARLETT2_SEGMENT_ID_FIRMWARE 1
+#define SCARLETT2_SEGMENT_ID_COUNT 2
+
+#define SCARLETT2_IOCTL_SELECT_FLASH_SEGMENT _IOW('S', 0x62, int)
+
+/* Erase selected flash segment */
+#define SCARLETT2_IOCTL_ERASE_FLASH_SEGMENT _IO('S', 0x63)
+
+/* Get selected flash segment erase progress
+ * 1 through to num_blocks, or 255 for complete
+ */
+struct scarlett2_flash_segment_erase_progress {
+ unsigned char progress;
+ unsigned char num_blocks;
+};
+#define SCARLETT2_IOCTL_GET_ERASE_PROGRESS \
+ _IOR('S', 0x64, struct scarlett2_flash_segment_erase_progress)
+
+#endif /* __UAPI_SOUND_SCARLETT2_H */
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index 453cab2a1209..ee5708934614 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -35,6 +35,7 @@
/* buffers */
#define SOF_TKN_BUF_SIZE 100
#define SOF_TKN_BUF_CAPS 101
+#define SOF_TKN_BUF_FLAGS 102
/* DAI */
/* Token retired with ABI 3.2, do not use for new capabilities
@@ -213,4 +214,8 @@
#define SOF_TKN_AMD_ACPI2S_CH 1701
#define SOF_TKN_AMD_ACPI2S_TDM_MODE 1702
+/* MICFIL PDM */
+#define SOF_TKN_IMX_MICFIL_RATE 2000
+#define SOF_TKN_IMX_MICFIL_CH 2001
+
#endif