From: Sangchul Lee <sc11.lee@samsung.com>
Date: Wed, 15 Sep 2021 09:07:53 +0000 (+0900)
Subject: webrtc_sink: Use fakesink to drop receiving audio data if stream_info is not set
X-Git-Tag: submit/tizen/20210916.061019^0
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=3624d1766927ad083acd2da0039a26a53d2fd9e7;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_sink: Use fakesink to drop receiving audio data if stream_info is not set

[Version] 0.2.100
[Issue Type] Improvement

Change-Id: I056d6b1b4fce1dfcb0a20a969631326ec7d7be7d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 4ecab4e8..f0737f9a 100644
--- a/packaging/capi-media-webrtc.spec
+++ b/packaging/capi-media-webrtc.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-webrtc
 Summary:    A WebRTC library in Tizen Native API
-Version:    0.2.99
+Version:    0.2.100
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_sink.c b/src/webrtc_sink.c
index 3afae365..585d9c2a 100644
--- a/src/webrtc_sink.c
+++ b/src/webrtc_sink.c
@@ -221,6 +221,7 @@ static int __build_audiosink(webrtc_s *webrtc, GstElement *decodebin, GstPad *sr
 	GstElement *audioconvert;
 	GstElement *audioresample;
 	GstElement *audiosink;
+	bool stream_info_is_set;
 	int ret = WEBRTC_ERROR_NONE;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
@@ -233,47 +234,62 @@ static int __build_audiosink(webrtc_s *webrtc, GstElement *decodebin, GstPad *sr
 
 	sink->media_types = MEDIA_TYPE_AUDIO;
 
-	if (!(audioconvert = _create_element(DEFAULT_ELEMENT_AUDIOCONVERT, NULL))) {
-		LOG_ERROR("failed to create audioconvert");
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
+	stream_info_is_set = !!sink->sound_stream_info.type;
 
-	if (!(audioresample = _create_element(DEFAULT_ELEMENT_AUDIORESAMPLE, NULL))) {
-		LOG_ERROR("failed to create audioresample");
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
+	if (stream_info_is_set) {
+		if (!(audioconvert = _create_element(DEFAULT_ELEMENT_AUDIOCONVERT, NULL)))
+			return WEBRTC_ERROR_INVALID_OPERATION;
 
-	if (!(audiosink = _create_element(webrtc->ini.rendering_sink.a_sink_element, NULL))) {
-		LOG_ERROR("failed to create audiosink");
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
+		if (!(audioresample = _create_element(DEFAULT_ELEMENT_AUDIORESAMPLE, NULL)))
+			return WEBRTC_ERROR_INVALID_OPERATION;
+
+		if (!(audiosink = _create_element(webrtc->ini.rendering_sink.a_sink_element, NULL)))
+			return WEBRTC_ERROR_INVALID_OPERATION;
 
-	if (g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(audiosink)), "stream-properties")) {
-		if (sink->sound_stream_info.type) {
-			ret = _apply_stream_info(audiosink, sink->sound_stream_info.type, sink->sound_stream_info.index);
-			if (ret != WEBRTC_ERROR_NONE) /* FIXME: unref all the created elements */
-				return WEBRTC_ERROR_INVALID_OPERATION;
+		if (g_object_class_find_property(G_OBJECT_GET_CLASS(G_OBJECT(audiosink)), "stream-properties")) {
+			if (sink->sound_stream_info.type) {
+				ret = _apply_stream_info(audiosink, sink->sound_stream_info.type, sink->sound_stream_info.index);
+				if (ret != WEBRTC_ERROR_NONE) /* FIXME: unref all the created elements */
+					return WEBRTC_ERROR_INVALID_OPERATION;
+			}
 		}
-	}
 
-	gst_bin_add_many(sink->bin, audioconvert, audioresample, audiosink, NULL);
+		gst_bin_add_many(sink->bin, audioconvert, audioresample, audiosink, NULL);
 
-	if (!gst_element_sync_state_with_parent(audioconvert)) {
-		LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audioconvert));
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
-	if (!gst_element_sync_state_with_parent(audioresample)) {
-		LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audioresample));
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
-	if (!gst_element_sync_state_with_parent(audiosink)) {
-		LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audiosink));
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
+		if (!gst_element_sync_state_with_parent(audioconvert)) {
+			LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audioconvert));
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+		if (!gst_element_sync_state_with_parent(audioresample)) {
+			LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audioresample));
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+		if (!gst_element_sync_state_with_parent(audiosink)) {
+			LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audiosink));
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
 
-	if (!gst_element_link_many(decodebin, audioconvert, audioresample, audiosink, NULL)) {
-		LOG_ERROR("failed to gst_element_link_many()");
-		return WEBRTC_ERROR_INVALID_OPERATION;
+		if (!gst_element_link_many(decodebin, audioconvert, audioresample, audiosink, NULL)) {
+			LOG_ERROR("failed to gst_element_link_many()");
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+
+	} else {
+		LOG_WARNING("stream_info is not set, use [%s] to drop audio data", DEFAULT_ELEMENT_FAKESINK);
+		if (!(audiosink = _create_element(DEFAULT_ELEMENT_FAKESINK, NULL)))
+			return WEBRTC_ERROR_INVALID_OPERATION;
+
+		gst_bin_add(sink->bin, audiosink);
+
+		if (!gst_element_sync_state_with_parent(audiosink)) {
+			LOG_ERROR("failed to gst_element_sync_state_with_parent() for [%s]", GST_ELEMENT_NAME(audiosink));
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+
+		if (!gst_element_link_many(decodebin, audiosink, NULL)) {
+			LOG_ERROR("failed to gst_element_link_many()");
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
 	}
 
 	return WEBRTC_ERROR_NONE;