From: Sangchul Lee <sc11.lee@samsung.com>
Date: Wed, 8 Jun 2022 06:59:36 +0000 (+0900)
Subject: webrtc_source: Save audio mute value if required element does not exist
X-Git-Tag: submit/tizen/20220613.225802~4
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=a92b2c01a90a7ee3324c8a8988b8c2f454caa21d;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_source: Save audio mute value if required element does not exist

This is a preparation for setting option values regardless of elements
creation or linking pads.

[Version] 0.3.120
[Issue Type] Improvement

Change-Id: I7747fe7df7a4e5e0cef0ffe21ccad1358bb27d4b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 3163f109..a93bd514 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:    0.3.119
+Version:    0.3.120
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index fed43ab3..eb2c5995 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -1532,8 +1532,14 @@ static int __build_rest_of_audiosrc(webrtc_s *webrtc, GstPad *src_pad, GstElemen
 
 	if (!(volume = _create_element(DEFAULT_ELEMENT_VOLUME, ELEMENT_NAME_VOLUME)))
 		goto exit;
+	if (!g_object_class_find_property(G_OBJECT_GET_CLASS(volume), "mute")) {
+		LOG_ERROR("there is no mute property");
+		goto exit;
+	}
 	APPEND_ELEMENT(element_list, volume);
 
+	g_object_set(volume, "mute", (gboolean)source->av[AV_IDX_AUDIO].mute, NULL);
+
 	source->av[AV_IDX_AUDIO].inbandfec = ini_source->use_inbandfec;
 	source->av[AV_IDX_AUDIO].packet_loss_percentage = ini_source->packet_loss_percentage;
 
@@ -3732,7 +3738,6 @@ int _set_video_mute(webrtc_s *webrtc, unsigned int source_id, bool mute)
 static int __validate_audio_source_for_mute(webrtc_s *webrtc, unsigned int source_id, webrtc_gst_slot_s **source)
 {
 	webrtc_gst_slot_s *_source;
-	GstElement *volume;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
@@ -3749,14 +3754,6 @@ static int __validate_audio_source_for_mute(webrtc_s *webrtc, unsigned int sourc
 		return WEBRTC_ERROR_INVALID_PARAMETER;
 	}
 
-	volume = gst_bin_get_by_name(_source->bin, ELEMENT_NAME_VOLUME);
-	RET_VAL_IF(volume == NULL, WEBRTC_ERROR_INVALID_OPERATION, "volume is NULL");
-
-	if (!g_object_class_find_property(G_OBJECT_GET_CLASS(volume), "mute")) {
-		LOG_ERROR("there is no mute property");
-		return WEBRTC_ERROR_INVALID_OPERATION;
-	}
-
 	if (source)
 		*source = _source;
 
@@ -3767,13 +3764,22 @@ int _set_audio_mute(webrtc_s *webrtc, unsigned int source_id, bool mute)
 {
 	int ret;
 	webrtc_gst_slot_s *source;
+	GstElement *volume;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
 	if ((ret = __validate_audio_source_for_mute(webrtc, source_id, &source)) != WEBRTC_ERROR_NONE)
 		return ret;
 
-	g_object_set(gst_bin_get_by_name(source->bin, ELEMENT_NAME_VOLUME), "mute", mute, NULL);
+	if ((volume = gst_bin_get_by_name(source->bin, ELEMENT_NAME_VOLUME))) {
+		if (!g_object_class_find_property(G_OBJECT_GET_CLASS(volume), "mute")) {
+			LOG_ERROR("there is no mute property");
+			return WEBRTC_ERROR_INVALID_OPERATION;
+		}
+		g_object_set(volume, "mute", mute, NULL);
+	}
+
+	source->av[AV_IDX_AUDIO].mute = mute;
 
 	LOG_INFO("webrtc[%p] source_id[%u] mute[%d]", webrtc, source_id, mute);
 
@@ -3804,7 +3810,6 @@ int _get_audio_mute(webrtc_s *webrtc, unsigned int source_id, bool *muted)
 {
 	int ret;
 	webrtc_gst_slot_s *source;
-	gboolean _muted;
 
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 	RET_VAL_IF(source_id == 0, WEBRTC_ERROR_INVALID_PARAMETER, "source_id is 0");
@@ -3812,9 +3817,7 @@ int _get_audio_mute(webrtc_s *webrtc, unsigned int source_id, bool *muted)
 	if ((ret = __validate_audio_source_for_mute(webrtc, source_id, &source)) != WEBRTC_ERROR_NONE)
 		return ret;
 
-	g_object_get(gst_bin_get_by_name(source->bin, ELEMENT_NAME_VOLUME), "mute", &_muted, NULL);
-
-	*muted = (bool)_muted;
+	*muted = source->av[AV_IDX_AUDIO].mute;
 
 	LOG_INFO("webrtc[%p] source_id[%u] muted[%d]", webrtc, source_id, *muted);