autochange - step2

marcinszkudlinski · Sep 12, 2024 · 027a4cc · 027a4cc
1 parent aa1e390
commit 027a4cc
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Showing 22 changed files with 33 additions and 33 deletions.
diff --git a/src/audio/asrc/asrc.c b/src/audio/asrc/asrc.c
@@ -396,7 +396,7 @@ static int asrc_params(struct processing_module *mod)
 		return -EINVAL;
 	}
 
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 	sinkb = comp_dev_get_first_data_consumer(dev);
 
 	/* update the source/sink buffer formats. Sink rate will be modified below */
@@ -543,7 +543,7 @@ static int asrc_prepare(struct processing_module *mod,
 		return ret;
 
 	/* SRC component will only ever have 1 source and 1 sink buffer */
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 	sinkb = comp_dev_get_first_data_consumer(dev);
 
 	/* get source data format and period bytes */
@@ -792,7 +792,7 @@ static int asrc_process(struct processing_module *mod,
 		return ret;
 
 	/* asrc component needs 1 source and 1 sink buffer */
-	source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	source = comp_dev_get_first_data_producer(dev);
 	sink = comp_dev_get_first_data_consumer(dev);
 
 	frames_src = audio_stream_get_avail_frames(source_s);

diff --git a/src/audio/codec/dts/dts.c b/src/audio/codec/dts/dts.c
@@ -76,7 +76,7 @@ static int dts_effect_convert_sof_interface_result(struct comp_dev *dev,
 static int dts_effect_populate_buffer_configuration(struct comp_dev *dev,
 	DtsSofInterfaceBufferConfiguration *buffer_config)
 {
-	struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	struct comp_buffer *source = comp_dev_get_first_data_producer(dev);
 	const struct audio_stream *stream;
 	DtsSofInterfaceBufferLayout buffer_layout;
 	DtsSofInterfaceBufferFormat buffer_format;

diff --git a/src/audio/dai-legacy.c b/src/audio/dai-legacy.c
@@ -493,7 +493,7 @@ int dai_common_params(struct dai_data *dd, struct comp_dev *dev,
 	}
 
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
-		dd->local_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		dd->local_buffer = comp_dev_get_first_data_producer(dev);
 	else
 		dd->local_buffer = comp_dev_get_first_data_consumer(dev);
 

diff --git a/src/audio/dai-zephyr.c b/src/audio/dai-zephyr.c
@@ -854,7 +854,7 @@ static int dai_set_dma_buffer(struct dai_data *dd, struct comp_dev *dev,
 	comp_dbg(dev, "dai_set_dma_buffer()");
 
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
-		dd->local_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		dd->local_buffer = comp_dev_get_first_data_producer(dev);
 	else
 		dd->local_buffer = comp_dev_get_first_data_consumer(dev);
 
@@ -1524,7 +1524,7 @@ static void set_new_local_buffer(struct dai_data *dd, struct comp_dev *dev)
 	uint32_t local_fmt;
 
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
-		dd->local_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		dd->local_buffer = comp_dev_get_first_data_producer(dev);
 	else
 		dd->local_buffer = comp_dev_get_first_data_consumer(dev);
 

diff --git a/src/audio/eq_iir/eq_iir_ipc3.c b/src/audio/eq_iir/eq_iir_ipc3.c
@@ -274,7 +274,7 @@ static int eq_iir_verify_params(struct comp_dev *dev,
 	comp_dbg(dev, "eq_iir_verify_params()");
 
 	/* EQ component will only ever have 1 source and 1 sink buffer */
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 	sinkb = comp_dev_get_first_data_consumer(dev);
 
 	/* we check whether we can support frame_fmt conversion (whether we have

diff --git a/src/audio/google/google_hotword_detect.c b/src/audio/google/google_hotword_detect.c
@@ -170,7 +170,7 @@ static int ghd_params(struct comp_dev *dev,
 	}
 
 	/* This detector component will only ever have 1 source */
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	if (audio_stream_get_channels(sourceb->stream) != 1) {
 		comp_err(dev, "ghd_params(): Only single-channel supported");
@@ -390,7 +390,7 @@ static int ghd_copy(struct comp_dev *dev)
 	}
 
 	/* keyword components will only ever have 1 source */
-	source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	source = comp_dev_get_first_data_producer(dev);
 	stream = &sourceb->stream;
 
 	bytes = audio_stream_get_avail_bytes(stream);

diff --git a/src/audio/google/google_rtc_audio_processing.c b/src/audio/google/google_rtc_audio_processing.c
@@ -633,7 +633,7 @@ static int google_rtc_audio_processing_prepare(struct processing_module *mod,
 	/* Don't need the ref buffer on IPC4 as pipelines are always
 	 * activated in tandem; also the API is deprecated
 	 */
-	cd->ref_comp_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	cd->ref_comp_buffer = comp_dev_get_first_data_producer(dev);
 	if (cd->aec_reference_source == 1)
 		cd->ref_comp_buffer = comp_dev_get_next_data_producer(dev, cd->ref_comp_buffer);
 #endif

diff --git a/src/audio/host-legacy.c b/src/audio/host-legacy.c
@@ -714,7 +714,7 @@ int host_common_params(struct host_data *hd, struct comp_dev *dev,
 	if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
 		hd->local_buffer = comp_dev_get_first_data_consumer(dev);
 	else
-		hd->local_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		hd->local_buffer = comp_dev_get_first_data_producer(dev);
 
 	period_bytes = dev->frames *
 		audio_stream_frame_bytes(&hd->local_buffer->stream);

diff --git a/src/audio/host-zephyr.c b/src/audio/host-zephyr.c
@@ -795,7 +795,7 @@ int host_common_params(struct host_data *hd, struct comp_dev *dev,
 	if (params->direction == SOF_IPC_STREAM_PLAYBACK)
 		hd->local_buffer = comp_dev_get_first_data_consumer(dev);
 	else
-		hd->local_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		hd->local_buffer = comp_dev_get_first_data_producer(dev);
 
 	period_bytes = dev->frames * get_frame_bytes(params->frame_fmt, params->channels);
 

diff --git a/src/audio/kpb.c b/src/audio/kpb.c
@@ -1188,7 +1188,7 @@ static int kpb_copy(struct comp_dev *dev)
 	}
 
 	/* Get source and sink buffers */
-	source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	source = comp_dev_get_first_data_producer(dev);
 
 	/* Validate source */
 	if (!audio_stream_get_rptr(&source->stream)) {

diff --git a/src/audio/module_adapter/module/waves/waves.c b/src/audio/module_adapter/module/waves/waves.c
@@ -216,7 +216,7 @@ static int waves_effect_allocate(struct processing_module *mod)
 static int waves_effect_check(struct comp_dev *dev)
 {
 	struct comp_buffer *sink = comp_dev_get_first_data_consumer(dev);
-	struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	struct comp_buffer *source = comp_dev_get_first_data_producer(dev);
 	const struct audio_stream *src_fmt = &source->stream;
 	const struct audio_stream *snk_fmt = &sink->stream;
 
@@ -282,7 +282,7 @@ static int waves_effect_check(struct comp_dev *dev)
 static int waves_effect_init(struct processing_module *mod)
 {
 	struct comp_dev *dev = mod->dev;
-	struct comp_buffer *source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	struct comp_buffer *source = comp_dev_get_first_data_producer(dev);
 	struct module_data *codec = &mod->priv;
 	struct waves_codec_data *waves_codec = codec->private;
 	const struct audio_stream *src_fmt = &source->stream;

diff --git a/src/audio/module_adapter/module_adapter_ipc3.c b/src/audio/module_adapter/module_adapter_ipc3.c
@@ -119,7 +119,7 @@ void module_adapter_check_data(struct processing_module *mod, struct comp_dev *d
 	 */
 	if (IS_PROCESSING_MODE_AUDIO_STREAM(mod) && mod->num_of_sources == 1 &&
 	    mod->num_of_sinks == 1) {
-		mod->source_comp_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		mod->source_comp_buffer = comp_dev_get_first_data_producer(dev);
 		mod->sink_comp_buffer = sink;
 		mod->stream_copy_single_to_single = true;
 	}

diff --git a/src/audio/module_adapter/module_adapter_ipc4.c b/src/audio/module_adapter/module_adapter_ipc4.c
@@ -219,7 +219,7 @@ static bool module_adapter_multi_sink_source_prepare(struct comp_dev *dev)
 
 	/* re-assign the source/sink modules */
 	mod->sink_comp_buffer = comp_dev_get_first_data_consumer(dev);
-	mod->source_comp_buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	mod->source_comp_buffer = comp_dev_get_first_data_producer(dev);
 
 	return false;
 }

diff --git a/src/audio/mux/mux_generic.c b/src/audio/mux/mux_generic.c
@@ -549,7 +549,7 @@ demux_func demux_get_processing_function(struct processing_module *mod)
 	if (list_is_empty(&dev->bsource_list))
 		return NULL;
 
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	for (i = 0; i < ARRAY_SIZE(mux_func_map); i++) {
 		enum sof_ipc_frame fmt = audio_stream_get_frm_fmt(&sourceb->stream);

diff --git a/src/audio/selector/selector.c b/src/audio/selector/selector.c
@@ -90,7 +90,7 @@ static int selector_verify_params(struct comp_dev *dev,
 		params->channels = out_channels;
 	} else {
 		/* fetch source buffer for capture */
-		buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		buffer = comp_dev_get_first_data_producer(dev);
 		if (cd->config.out_channels_count &&
 		    cd->config.out_channels_count != params->channels) {
 			comp_err(dev, "selector_verify_params(): src in_channels_count does not match pcm channels");
@@ -352,7 +352,7 @@ static int selector_trigger(struct comp_dev *dev, int cmd)
 
 	comp_dbg(dev, "selector_trigger()");
 
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	ret = comp_set_state(dev, cmd);
 
@@ -381,7 +381,7 @@ static int selector_copy(struct comp_dev *dev)
 	comp_dbg(dev, "selector_copy()");
 
 	/* selector component will have 1 source and 1 sink buffer */
-	source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	source = comp_dev_get_first_data_producer(dev);
 	sink = comp_dev_get_first_data_consumer(dev);
 
 	if (!audio_stream_get_avail(&source->stream))
@@ -428,7 +428,7 @@ static int selector_prepare(struct comp_dev *dev)
 		return PPL_STATUS_PATH_STOP;
 
 	/* selector component will have 1 source and 1 sink buffer */
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 	sinkb = comp_dev_get_first_data_consumer(dev);
 
 	/* get source data format and period bytes */

diff --git a/src/audio/tone.c b/src/audio/tone.c
@@ -428,7 +428,7 @@ static int tone_params(struct comp_dev *dev,
 	struct comp_data *cd = comp_get_drvdata(dev);
 	struct comp_buffer *sourceb, *sinkb;
 
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	sinkb = comp_dev_get_first_data_consumer(dev);
 
@@ -672,7 +672,7 @@ static int tone_prepare(struct comp_dev *dev)
 	if (ret == COMP_STATUS_STATE_ALREADY_SET)
 		return PPL_STATUS_PATH_STOP;
 
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	cd->channels = audio_stream_get_channels(&sourceb->stream);
 	comp_info(dev, "tone_prepare(), cd->channels = %u, cd->rate = %u",

diff --git a/src/audio/volume/volume.c b/src/audio/volume/volume.c
@@ -685,7 +685,7 @@ static int volume_prepare(struct processing_module *mod,
 
 	/* volume component will only ever have 1 sink and source buffer */
 	sinkb = comp_dev_get_first_data_consumer(dev);
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 
 	volume_set_alignment(&sourceb->stream, &sinkb->stream);
 

diff --git a/src/audio/volume/volume_ipc3.c b/src/audio/volume/volume_ipc3.c
@@ -39,7 +39,7 @@ void set_volume_process(struct vol_data *cd, struct comp_dev *dev, bool source_o
 	struct comp_buffer *bufferb;
 
 	if (source_or_sink)
-		bufferb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		bufferb = comp_dev_get_first_data_producer(dev);
 	else
 		bufferb = comp_dev_get_first_data_consumer(dev);
 

diff --git a/src/samples/audio/detect_test.c b/src/samples/audio/detect_test.c
@@ -818,7 +818,7 @@ static int test_keyword_params(struct comp_dev *dev,
 	cd->sample_valid_bytes = params->sample_valid_bytes;
 
 	/* keyword components will only ever have 1 source */
-	sourceb = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	sourceb = comp_dev_get_first_data_producer(dev);
 	channels = audio_stream_get_channels(&sourceb->stream);
 	frame_fmt = audio_stream_get_frm_fmt(&sourceb->stream);
 	rate = audio_stream_get_rate(&sourceb->stream);
@@ -894,7 +894,7 @@ static int test_keyword_copy(struct comp_dev *dev)
 	comp_dbg(dev, "test_keyword_copy()");
 
 	/* keyword components will only ever have 1 source */
-	source = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	source = comp_dev_get_first_data_producer(dev);
 
 	if (!audio_stream_get_avail(&source->stream))
 		return PPL_STATUS_PATH_STOP;

diff --git a/tools/plugin/modules/alsa.c b/tools/plugin/modules/alsa.c
@@ -460,7 +460,7 @@ static int aplay_params(struct comp_dev *dev, struct sof_ipc_stream_params *para
 	memcpy(&cd->params, params, sizeof(*params));
 
 	/* file component sink/source buffer period count */
-	buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	buffer = comp_dev_get_first_data_producer(dev);
 	buffer_reset_pos(buffer, NULL);
 
 	comp_dbg(dev, "prepare done ret = %d", ret);
@@ -586,7 +586,7 @@ static int aplay_copy(struct comp_dev *dev)
 	}
 
 	/* file component source buffer */
-	buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	buffer = comp_dev_get_first_data_producer(dev);
 	source = &buffer->stream;
 	pos = source->r_ptr;
 	avail = audio_stream_get_avail_frames(source);

diff --git a/tools/plugin/modules/shm.c b/tools/plugin/modules/shm.c
@@ -247,7 +247,7 @@ static int shmread_copy(struct comp_dev *dev)
 	void *dest;
 
 	/* local SOF source buffer */
-	buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+	buffer = comp_dev_get_first_data_producer(dev);
 	source = &buffer->stream;
 	rptr = source->r_ptr;
 

diff --git a/tools/testbench/file.c b/tools/testbench/file.c
@@ -829,7 +829,7 @@ static int file_copy(struct comp_dev *dev)
 		break;
 	case FILE_WRITE:
 		/* file component source buffer */
-		buffer = list_first_item(&dev->bsource_list, struct comp_buffer, Xsink_list);
+		buffer = comp_dev_get_first_data_producer(dev);
 
 		/* test source has enough free frames */
 		src_frames = audio_stream_get_avail_frames(&buffer->stream);