From: Sangchul Lee <sc11.lee@samsung.com>
Date: Thu, 3 Jun 2021 08:50:28 +0000 (+0900)
Subject: webrtc_private: Set latency to webrtcbin directly
X-Git-Tag: submit/tizen/20210729.023123~61
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=refs%2Fchanges%2F67%2F259267%2F2;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_private: Set latency to webrtcbin directly

The webrtcbin based on 1.18 version has latency property that
set the latency property of rtpbin internally.

[Version] 0.2.2
[Issue Type] Update

Change-Id: I18716dacc202ad195b844365e9e8c21300d1d121
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 2fd644b9..c7eeb067 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:    0.2.1
+Version:    0.2.2
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_private.c b/src/webrtc_private.c
index 6a826fe8..24226ae4 100644
--- a/src/webrtc_private.c
+++ b/src/webrtc_private.c
@@ -1000,8 +1000,6 @@ static void __webrtcbin_on_new_transceiver_cb(GstElement *webrtcbin, GstWebRTCRT
 
 int _gst_build_pipeline(webrtc_s *webrtc)
 {
-	GstElement *rtpbin;
-
 	RET_VAL_IF(webrtc == NULL, WEBRTC_ERROR_INVALID_PARAMETER, "webrtc is NULL");
 
 	webrtc->gst.pipeline = gst_pipeline_new("webrtc-pipeline");
@@ -1030,12 +1028,7 @@ int _gst_build_pipeline(webrtc_s *webrtc)
 		LOG_INFO("stun_server[%s]", webrtc->stun_server_url);
 	}
 
-	if (!(rtpbin = gst_bin_get_by_name(GST_BIN(webrtc->gst.webrtcbin), "rtpbin"))) {
-		LOG_ERROR("failed to get rtpbin");
-		goto error;
-	}
-	g_object_set(G_OBJECT(rtpbin), "latency", webrtc->ini.general.jitterbuffer_latency, NULL);
-	gst_object_unref(rtpbin);
+	g_object_set(G_OBJECT(webrtc->gst.webrtcbin), "latency", webrtc->ini.general.jitterbuffer_latency, NULL);
 
 	_connect_and_append_signal(&webrtc->signals, (GObject *)webrtc->gst.webrtcbin, "on-negotiation-needed", G_CALLBACK(__webrtcbin_on_negotiation_needed_cb), webrtc);
 	_connect_and_append_signal(&webrtc->signals, (GObject *)webrtc->gst.webrtcbin, "on-ice-candidate", G_CALLBACK(__webrtcbin_on_ice_candidate_cb), webrtc);