Merge "ASoC: msm: qdsp6v2: Support to set session start delay"

3 files changed, 128 insertions, 2 deletions

diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index ddc21d0c1bbb..4bfd187095b6 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -10224,12 +10224,108 @@ 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