google_rtc_audio_processing: Add config controls

src/audio/google_rtc_audio_processing.c

@@ -84,6 +84,39 @@ static int google_rtc_audio_processing_params(
 	return 0;
 }
 
+static int google_rtc_audio_processing_reconfigure(struct comp_dev *dev)
+{
+	struct google_rtc_audio_processing_comp_data *cd = comp_get_drvdata(dev);
+	uint8_t *config;
+	size_t size;
+	int ret;
+
+	comp_dbg(dev, "google_rtc_audio_processing_reconfigure()");
+
+	config = comp_get_data_blob(cd->tuning_handler, &size, NULL);
+	if (size == 0) {
+		/* No data to be handled */
+		return 0;
+	}
+
+	if (!config) {
+		comp_err(dev, "google_rtc_audio_processing_reconfigure(): Tuning config not set");
+		return -EINVAL;
+	}
+
+	comp_info(dev, "google_rtc_audio_processing_reconfigure(): New tuning config %p (%zu bytes)",
+		  config, size);
+
+	ret = GoogleRtcAudioProcessingReconfigure(cd->state, config, size);
+	if (ret) {
+		comp_err(dev, "GoogleRtcAudioProcessingReconfigure failed: %d",
+			 ret);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static int google_rtc_audio_processing_cmd_set_data(
 	struct comp_dev *dev,
 	struct sof_ipc_ctrl_data *cdata)
@@ -129,6 +162,9 @@ static int google_rtc_audio_processing_cmd(struct comp_dev *dev, int cmd, void *
 	comp_dbg(dev, "google_rtc_audio_processing_cmd(): %d - data_cmd: %d", cmd, cdata->cmd);
 
 	switch (cmd) {
+	case COMP_CMD_SET_VALUE:
+	case COMP_CMD_GET_VALUE:
+		return 0;
 	case COMP_CMD_SET_DATA:
 		return google_rtc_audio_processing_cmd_set_data(dev, cdata);
 	case COMP_CMD_GET_DATA:
@@ -246,6 +282,7 @@ static int google_rtc_audio_processing_prepare(struct comp_dev *dev)
 	struct google_rtc_audio_processing_comp_data *cd = comp_get_drvdata(dev);
 	struct comp_buffer *source_buffer;
 	struct list_item *source_buffer_list_item;
+	int ret;
 
 	comp_dbg(dev, "google_rtc_audio_processing_prepare()");
 
@@ -288,6 +325,13 @@ static int google_rtc_audio_processing_prepare(struct comp_dev *dev)
 		return -EINVAL;
 	}
 
+	/* Blobs sent during COMP_STATE_READY is assigned to blob_handler->data
+	 * directly, so comp_is_new_data_blob_available always returns false.
+	 */
+	ret = google_rtc_audio_processing_reconfigure(dev);
+	if (ret)
+		return ret;
+
 	return comp_set_state(dev, COMP_TRIGGER_PREPARE);
 }
 
@@ -312,6 +356,13 @@ static int google_rtc_audio_processing_copy(struct comp_dev *dev)
 	uint32_t num_aec_reference_frames;
 	uint32_t num_aec_reference_bytes;
 	int channel;
+	int ret;
+
+	if (comp_is_new_data_blob_available(cd->tuning_handler)) {
+		ret = google_rtc_audio_processing_reconfigure(dev);
+		if (ret)
+			return ret;
+	}
 
 	cd->aec_reference = buffer_acquire(cd->aec_reference);
 	num_aec_reference_frames = audio_stream_get_avail_frames(&cd->aec_reference->stream);

src/audio/google_rtc_audio_processing_mock.c

@@ -31,11 +31,11 @@ GoogleRtcAudioProcessingState *GoogleRtcAudioProcessingCreate()
 	s->num_output_channels = 1;
 	s->num_frames = GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ * 10 / 1000;
 	s->aec_reference = rballoc(0,
-							   SOF_MEM_CAPS_RAM,
-							   sizeof(s->aec_reference[0]) *
-							   s->num_frames *
-							   s->num_aec_reference_channels
-							   );
+				   SOF_MEM_CAPS_RAM,
+				   sizeof(s->aec_reference[0]) *
+				   s->num_frames *
+				   s->num_aec_reference_channels
+				   );
 	return s;
 }
 
@@ -52,6 +52,13 @@ int GoogleRtcAudioProcessingGetFramesizeInMs(GoogleRtcAudioProcessingState *stat
 	return state->num_frames * 1000 / GOOGLE_RTC_AUDIO_PROCESSING_SAMPLE_RATE_HZ;
 }
 
+int GoogleRtcAudioProcessingReconfigure(
+	GoogleRtcAudioProcessingState * const state, const uint8_t *const config,
+	int config_size)
+{
+	return 0;
+}
+
 int GoogleRtcAudioProcessingProcessCapture_int16(
 	GoogleRtcAudioProcessingState *const state, const int16_t *const src,
 	int16_t *const dest)
@@ -78,6 +85,6 @@ int GoogleRtcAudioProcessingAnalyzeRender_int16(
 			* state->num_frames
 			* state->num_aec_reference_channels;
 	memcpy_s(state->aec_reference, buffer_size,
-			 data, buffer_size);
+		 data, buffer_size);
 	return 0;
 }

third_party/include/google_rtc_audio_processing.h

@@ -26,38 +26,44 @@ void GoogleRtcAudioProcessingFree(GoogleRtcAudioProcessingState *state);
 
 // Returns the framesize used for processing.
 int GoogleRtcAudioProcessingGetFramesizeInMs(
-	GoogleRtcAudioProcessingState *state);
+    GoogleRtcAudioProcessingState *const state);
+
+// Reconfigure the audio processing.
+// Returns 0 if success and non zero if failure.
+int GoogleRtcAudioProcessingReconfigure(
+    GoogleRtcAudioProcessingState *const state, const uint8_t *const config,
+    int config_size);
 
 // Processes the microphone stream.
 // Accepts deinterleaved float audio with the range [-1, 1]. Each element of
 // |src| points to an array of samples for the channel. At output, the channels
 // will be in |dest|.
 // Returns 0 if success and non zero if failure.
 int GoogleRtcAudioProcessingProcessCapture_float32(
-	GoogleRtcAudioProcessingState *const state, const float *const *src,
-	float *const *dest);
+    GoogleRtcAudioProcessingState *const state, const float *const *src,
+    float *const *dest);
 
 // Accepts and and produces a frame of interleaved 16 bit integer audio. `src`
 // and `dest` may use the same memory, if desired.
 // Returns 0 if success and non zero if failure.
 int GoogleRtcAudioProcessingProcessCapture_int16(
-	GoogleRtcAudioProcessingState *const state, const int16_t *const src,
-	int16_t *const dest);
+    GoogleRtcAudioProcessingState *const state, const int16_t *const src,
+    int16_t *const dest);
 
 // Analyzes the playback stream.
 // Accepts deinterleaved float audio with the range [-1, 1]. Each element
 // of |src| points to an array of samples for the channel.
 // Returns 0 if success and non zero if failure.
 int GoogleRtcAudioProcessingAnalyzeRender_float32(
-	GoogleRtcAudioProcessingState *const state, const float *const *src);
+    GoogleRtcAudioProcessingState *const state, const float *const *src);
 
 // Accepts interleaved int16 audio.
 // Returns 0 if success and non zero if failure.
 int GoogleRtcAudioProcessingAnalyzeRender_int16(
-	GoogleRtcAudioProcessingState *const state, const int16_t *const src);
+    GoogleRtcAudioProcessingState *const state, const int16_t *const src);
 
 #ifdef __cplusplus
 }
 #endif
 
-#endif	// GOOGLE_RTC_AUDIO_PROCESSING_H
+#endif  // GOOGLE_RTC_AUDIO_PROCESSING_H

tools/topology/topology1/m4/google_rtc_audio_processing.m4

@@ -7,7 +7,7 @@ DECLARE_SOF_RT_UUID("google-rtc-audio-processing", google_rtc_audio_processing_u
 dnl N_GOOGLE_RTC_AUDIO_PROCESSING(name)
 define(`N_GOOGLE_RTC_AUDIO_PROCESSING', `GOOGLE_RTC_AUDIO_PROCESSING'PIPELINE_ID`.'$1)
 
-dnl W_GOOGLE_RTC_AUDIO_PROCESSING(name, format, periods_sink, periods_source, core, kcontrols_list)
+dnl W_GOOGLE_RTC_AUDIO_PROCESSING(name, format, periods_sink, periods_source, core, mixer_kcontrols_list, byte_kcontrols_list)
 define(`W_GOOGLE_RTC_AUDIO_PROCESSING',
 `SectionVendorTuples."'N_GOOGLE_RTC_AUDIO_PROCESSING($1)`_tuples_uuid" {'
 `	tokens "sof_comp_tokens"'
@@ -57,9 +57,12 @@ define(`W_GOOGLE_RTC_AUDIO_PROCESSING',
 `		"'N_GOOGLE_RTC_AUDIO_PROCESSING($1)`_data_str"'
 `		"'N_GOOGLE_RTC_AUDIO_PROCESSING($1)`_data_str_type"'
 `	]'
-`	bytes ['
+`	mixer ['
 		$6
 `	]'
+`	bytes ['
+		$7
+`	]'
 `}')
 
 divert(0)dnl

tools/topology/topology1/m4/google_rtc_audio_processing_default.m4

@@ -1,20 +1,10 @@
+# Google RTC Audio Processing initial config.
+# Empty configuration
+# sof_abi_hdr.size = 0
+# sof_abi_hdr.type = 0 (BINARY_DATA)
 CONTROLBYTES_PRIV(GOOGLE_RTC_AUDIO_PROCESSING_priv,
 `       bytes "0x53,0x4f,0x46,0x00,0x00,0x00,0x00,0x00,'
-`       0x6c,0x00,0x00,0x00,0x00,0x20,0x01,0x03,'
+`       0x00,0x00,0x00,0x00,0x00,0x50,0x01,0x03,'
 `       0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,'
-`       0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,'
-`       0x6c,0x00,0x00,0x00,0x00,0x00,0x00,0x00,'
-`       0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,'
-`       0x00,0x00,0x00,0x00,0x01,0x00,0x00,0x00,'
-`       0x00,0x00,0x00,0xe8,0x00,0x00,0x00,0x1e,'
-`       0x00,0x00,0x00,0x01,0xd3,0x4d,0x62,0x00,'
-`       0xb1,0xc2,0x09,0x04,0x00,0x00,0x00,0x40,'
-`       0xa6,0x99,0x01,0x00,0xce,0xd8,0x0f,0x0a,'
-`       0x1f,0x1a,0xf0,0xf5,0x83,0xc9,0xfe,0x01,'
-`       0xdf,0x30,0x61,0x3a,0xcb,0x83,0x0e,0x01,'
-`       0xef,0x84,0x73,0x01,0x6d,0x64,0x6b,0xff,'
-`       0x03,0x41,0x22,0x00,0x05,0x00,0x00,0x00,'
-`       0x00,0x10,0xf8,0x00,0xaa,0x81,0x10,0x00,'
-`       0xf4,0xf0,0x8a,0x00,0xaa,0xe1,0x22,0x00,'
-`       0xb9,0x9b,0x02,0x00"'
+`       0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00"'
 )

tools/topology/topology1/sof/pipe-google-rtc-audio-processing-capture.m4

@@ -16,16 +16,20 @@ include(`bytecontrol.m4')
 include(`enumcontrol.m4')
 include(`google_rtc_audio_processing.m4')
 
-define(GOOGLE_RTC_AUDIO_PROCESSING_priv, concat(`google_rtc_audio_processing_bytes_', PIPELINE_ID))
-define(GOOGLE_RTC_AUDIO_PROCESSING_CTRL, concat(`google_rtc_audio_processing_control_', PIPELINE_ID))
+#
+# Controls
+#
+
 include(`google_rtc_audio_processing_default.m4')
-C_CONTROLBYTES(GOOGLE_RTC_AUDIO_PROCESSING_CTRL, PIPELINE_ID,
-      CONTROLBYTES_OPS(bytes, 258 binds the control to bytes get/put handlers, 258, 258),
-      CONTROLBYTES_EXTOPS(258 binds the control to bytes get/put handlers, 258, 258),
-      , , ,
-      CONTROLBYTES_MAX(, 2048),
-      ,
-      GOOGLE_RTC_AUDIO_PROCESSING_priv)
+
+# Byte control for AEC tuning data
+C_CONTROLBYTES(`Config', PIPELINE_ID,
+	CONTROLBYTES_OPS(bytes, 258 binds the control to bytes get/put handlers, 258, 258),
+	CONTROLBYTES_EXTOPS(258 binds the control to bytes get/put handlers, 258, 258),
+	, , ,
+	CONTROLBYTES_MAX(, 2048),
+	,
+	GOOGLE_RTC_AUDIO_PROCESSING_priv)
 
 #
 # Components and Buffers
@@ -36,7 +40,8 @@ C_CONTROLBYTES(GOOGLE_RTC_AUDIO_PROCESSING_CTRL, PIPELINE_ID,
 W_PCM_CAPTURE(PCM_ID, Google RTC Audio Processing, 0, DAI_PERIODS, SCHEDULE_CORE)
 
 W_GOOGLE_RTC_AUDIO_PROCESSING(0, PIPELINE_FORMAT, 2, DAI_PERIODS, SCHEDULE_CORE,
-			LIST(`          ', "GOOGLE_RTC_AUDIO_PROCESSING_CTRL"))
+	`',
+	LIST(`          ', "Config"))
 
 # Capture Buffers
 W_BUFFER(0, COMP_BUFFER_SIZE(2,