diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/linux/clocksource.h | 1 | ||||
| -rw-r--r-- | include/linux/mmc/host.h | 2 | ||||
| -rw-r--r-- | include/sound/apr_audio-v2.h | 101 | ||||
| -rw-r--r-- | include/sound/q6asm-v2.h | 8 | ||||
| -rw-r--r-- | include/uapi/sound/compress_offload.h | 22 |
5 files changed, 131 insertions, 3 deletions
diff --git a/include/linux/clocksource.h b/include/linux/clocksource.h index 7784b597e959..39c7de8c3048 100644 --- a/include/linux/clocksource.h +++ b/include/linux/clocksource.h @@ -186,6 +186,7 @@ extern void clocksource_suspend(void); extern void clocksource_resume(void); extern struct clocksource * __init clocksource_default_clock(void); extern void clocksource_mark_unstable(struct clocksource *cs); +extern void clocksource_select_force(void); extern u64 clocks_calc_max_nsecs(u32 mult, u32 shift, u32 maxadj, u64 mask, u64 *max_cycles); diff --git a/include/linux/mmc/host.h b/include/linux/mmc/host.h index 5374bd6c4cbe..055b879dfa6b 100644 --- a/include/linux/mmc/host.h +++ b/include/linux/mmc/host.h @@ -534,6 +534,8 @@ struct mmc_host { struct dentry *debugfs_root; + bool err_occurred; + struct mmc_async_req *areq; /* active async req */ struct mmc_context_info context_info; /* async synchronization info */ diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index ddc21d0c1bbb..06c273252484 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -933,7 +933,6 @@ struct adm_cmd_connect_afe_port_v5 { #define SLIMBUS_3_TX 0x4007 #define SLIMBUS_4_RX 0x4008 #define SLIMBUS_4_TX 0x4009 -#define SLIMBUS_TX_VI 0x4f09 #define SLIMBUS_5_RX 0x400a #define SLIMBUS_5_TX 0x400b #define SLIMBUS_6_RX 0x400c @@ -10224,12 +10223,108 @@ struct asm_session_cmd_set_mtmx_strstr_params_v2 { */ }; +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the + * audio client choose the rendering decision that the audio DSP should use. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_MODE_CMD 0x00012F0D + +/* Indicates that rendering decision will be based on default rate + * (session clock based rendering, device driven). + * 1. The default session clock based rendering is inherently driven + * by the timing of the device. + * 2. After the initial decision is made (first buffer after a run + * command), subsequent data rendering decisions are made with + * respect to the rate at which the device is rendering, thus deriving + * its timing from the device. + * 3. While this decision making is simple, it has some inherent limitations + * (mentioned in the next section). + * 4. If this API is not set, the session clock based rendering will be assumed + * and this will ensure that the DSP is backward compatible. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT 0 + +/* Indicates that rendering decision will be based on local clock rate. + * 1. In the DSP loopback/client loopback use cases (frame based + * inputs), the incoming data into audio DSP is time-stamped at the + * local clock rate (STC). + * 2. This TS rate may match the incoming data rate or maybe different + * from the incoming data rate. + * 3. Regardless, the data will be time-stamped with local STC and + * therefore, the client is recommended to set this mode for these + * use cases. This method is inherently more robust to sequencing + * (AFE Start/Stop) and device switches, among other benefits. + * 4. This API will inform the DSP to compare every incoming buffer TS + * against local STC. + * 5. DSP will continue to honor render windows APIs, as before. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC 1 + +/* Structure for rendering decision parameter */ +struct asm_session_mtmx_strtr_param_render_mode_t { + /* Specifies the type of rendering decision the audio DSP should use. + * + * @values + * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT + * - #ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC + */ + u32 flags; +} __packed; + +/* Parameter used by #ASM_SESSION_MTMX_STRTR_MODULE_ID_AVSYNC which allows the + * audio client to specify the clock recovery mechanism that the audio DSP + * should use. + */ + +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_CMD 0x00012F0E + +/* Indicates that default clock recovery will be used (no clock recovery). + * If the client wishes that no clock recovery be done, the client can + * choose this. This means that no attempt will made by the DSP to try and + * match the rates of the input and output audio. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE 0 + +/* Indicates that independent clock recovery needs to be used. + * 1. In the DSP loopback/client loopback use cases (frame based inputs), + * the client should choose the independent clock recovery option. + * 2. This basically de-couples the audio and video from knowing each others + * clock sources and lets the audio DSP independently rate match the input + * and output rates. + * 3. After drift detection, the drift correction is achieved by either pulling + * the PLLs (if applicable) or by stream to device rate matching + * (for PCM use cases) by comparing drift with respect to STC. + * 4. For passthrough use cases, since the PLL pulling is the only option, + * a best effort will be made. + * If PLL pulling is not possible / available, the rendering will be + * done without rate matching. + */ +#define ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO 1 + +/* Payload of the #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC parameter. + */ +struct asm_session_mtmx_strtr_param_clk_rec_t { + /* Specifies the type of clock recovery that the audio DSP should + * use for rate matching. + */ + + /* @values + * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_DEFAULT + * #ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_INDEPENDENT + */ + u32 flags; +} __packed; + +union asm_session_mtmx_strtr_param_config { + struct asm_session_mtmx_strtr_param_window_v2_t window_param; + struct asm_session_mtmx_strtr_param_render_mode_t render_param; + struct asm_session_mtmx_strtr_param_clk_rec_t clk_rec_param; +} __packed; + struct asm_mtmx_strtr_params { struct apr_hdr hdr; struct asm_session_cmd_set_mtmx_strstr_params_v2 param; struct asm_stream_param_data_v2 data; - u32 window_lsw; - u32 window_msw; + union asm_session_mtmx_strtr_param_config config; } __packed; #define ASM_SESSION_CMD_GET_MTMX_STRTR_PARAMS_V2 0x00010DCF diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h index 76bb795119c2..4947c30287a3 100644 --- a/include/sound/q6asm-v2.h +++ b/include/sound/q6asm-v2.h @@ -642,6 +642,14 @@ int q6asm_send_mtmx_strtr_window(struct audio_client *ac, struct asm_session_mtmx_strtr_param_window_v2_t *window_param, uint32_t param_id); +/* Configure DSP render mode */ +int q6asm_send_mtmx_strtr_render_mode(struct audio_client *ac, + uint32_t render_mode); + +/* Configure DSP clock recovery mode */ +int q6asm_send_mtmx_strtr_clk_rec_mode(struct audio_client *ac, + uint32_t clk_rec_mode); + /* Retrieve the current DSP path delay */ int q6asm_get_path_delay(struct audio_client *ac); diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index e050bc758b3b..30481056cce1 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -132,20 +132,42 @@ struct snd_compr_audio_info { __u32 reserved[15]; } __attribute__((packed, aligned(4))); +#define SNDRV_COMPRESS_RENDER_MODE_AUDIO_MASTER 0 +#define SNDRV_COMPRESS_RENDER_MODE_STC_MASTER 1 + +#define SNDRV_COMPRESS_CLK_REC_MODE_NONE 0 +#define SNDRV_COMPRESS_CLK_REC_MODE_AUTO 1 + /** * enum sndrv_compress_encoder * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the * end of the track * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the * beginning of the track + * @SNDRV_COMPRESS_PATH_DELAY: dsp path delay in microseconds + * @SNDRV_COMPRESS_RENDER_MODE: dsp render mode (audio master or stc) + * @SNDRV_COMPRESS_CLK_REC_MODE: clock recovery mode ( none or auto) + * @SNDRV_COMPRESS_RENDER_WINDOW: render window + * @SNDRV_COMPRESS_START_DELAY: start delay */ enum sndrv_compress_encoder { SNDRV_COMPRESS_ENCODER_PADDING = 1, SNDRV_COMPRESS_ENCODER_DELAY = 2, SNDRV_COMPRESS_MIN_BLK_SIZE = 3, SNDRV_COMPRESS_MAX_BLK_SIZE = 4, + SNDRV_COMPRESS_PATH_DELAY = 5, + SNDRV_COMPRESS_RENDER_MODE = 6, + SNDRV_COMPRESS_CLK_REC_MODE = 7, + SNDRV_COMPRESS_RENDER_WINDOW = 8, + SNDRV_COMPRESS_START_DELAY = 9, }; +#define SNDRV_COMPRESS_PATH_DELAY SNDRV_COMPRESS_PATH_DELAY +#define SNDRV_COMPRESS_RENDER_MODE SNDRV_COMPRESS_RENDER_MODE +#define SNDRV_COMPRESS_CLK_REC_MODE SNDRV_COMPRESS_CLK_REC_MODE +#define SNDRV_COMPRESS_RENDER_WINDOW SNDRV_COMPRESS_RENDER_WINDOW +#define SNDRV_COMPRESS_START_DELAY SNDRV_COMPRESS_START_DELAY + /** * struct snd_compr_metadata - compressed stream metadata * @key: key id |