webrtc_source: Disable clock synchronization of loopback pipeline audiosink

Webrtc handle can have a source that consists of audio, video or both
media types. Each type can have a loopback pipeline. So it is set to FALSE
to render the incoming data from pad probe callback as soon as possible.

[Version] 0.2.120
[Issue Type] Improvement

Change-Id: I3f46e96123031598a5d86fa8fc85c1ec96772e4a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index f3d572d664deedb9b7567403b69bfb35c5d7533f..02acb1d8492c06dd8bda38705355fc1be29c173a 100644 (file)
--- 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:    0.2.119
+Version:    0.2.120
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index a86355eb7d5597b8850804f4c8a7ffd99a53e8c1..a289ac35afcfc1f5aead481a10589ab4175d17ca 100644 (file)
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -3732,6 +3732,7 @@ static int __build_loopback_audiosink(webrtc_gst_slot_s *source, GstElement *lin
                        }
                }
        }
+       g_object_set(G_OBJECT(audiosink), "sync", FALSE, NULL);
 
        if (!(audioconvert = _create_element(DEFAULT_ELEMENT_AUDIOCONVERT, NULL))) {
                SAFE_GST_OBJECT_UNREF(audiosink);
@@ -3827,7 +3828,7 @@ static int __build_loopback_videosink(webrtc_gst_slot_s *source, GstElement *lin
                        NULL);
 
        } else if (source->display->type == WEBRTC_DISPLAY_TYPE_EVAS) {
-               g_object_set(videosink, "signal-handoffs", TRUE, NULL);
+               g_object_set(G_OBJECT(videosink), "signal-handoffs", TRUE, NULL);
                _connect_and_append_signal(&source->signals, (GObject *)videosink, "handoff", G_CALLBACK(_video_stream_decoded_cb), source->display);
        }