webrtc_transceiver: Revise the postfix number of mid

Considering the postfix value of gstreamer element name accumulates
even if the handle is destroyed and re-created in the same process,
it has been revised.

[Version] 0.4.51
[Issue Type] Bug fix

Change-Id: Ia513a40637230fcf158f7f8a8e584af104c9bd6e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 213c2133f986e383bbfd0f211e024913c0100017..f8f0a0779b273d62d2aff1e0a3f89aab9d7deabf 100644 (file)
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -574,6 +574,7 @@ typedef struct _webrtc_s {
        webrtc_rotate_info_s rotate_info;
 #endif
 #endif
+       unsigned int transceiver_count;
 } webrtc_s;
 
 /* FIXME: divide into two slot types or use union */

diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 4d1d9343ba3cda7a357d0c0ddb705d3fcb346edf..798b3f23c55d0bd1e5cc0b6b05ee7f10f7f06442 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:    0.4.50
+Version:    0.4.51
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/webrtc_transceiver.c b/src/webrtc_transceiver.c
index a8d75e524724c91c76cabfe05f47f3462e4da0de..94366001d2fdb74de602325f0b616f21f09e24ed 100644 (file)
--- a/src/webrtc_transceiver.c
+++ b/src/webrtc_transceiver.c
@@ -75,14 +75,9 @@ static void __webrtcbin_transceiver_set_fec_percentage(webrtc_s *webrtc, GstWebR
        LOG_INFO("set fec-percentage[%u] to transceiver[%p]", fec_percentage, transceiver);
 }
 
-static gchar *__get_gst_mid_from_transceiver(bool is_audio, const char *transceiver_name)
+static gchar *__get_mid_from_transceiver_count(bool is_audio, unsigned int count)
 {
-       g_auto(GStrv) str_arr = NULL;
-
-       RET_VAL_IF(transceiver_name == NULL, NULL, "transceiver_name is NULL");
-
-       str_arr = g_strsplit(transceiver_name, "webrtctransceiver", 2);
-       return g_strdup_printf("%s%s", is_audio ? "audio" : "video", str_arr[1]);
+       return g_strdup_printf("%s%u", is_audio ? "audio" : "video", count);
 }
 
 void _webrtcbin_on_new_transceiver_cb(GstElement *webrtcbin, GstWebRTCRTPTransceiver *transceiver, gpointer user_data)
@@ -133,7 +128,7 @@ void _webrtcbin_on_new_transceiver_cb(GstElement *webrtcbin, GstWebRTCRTPTransce
 
                        source->av[j].transceiver = gst_object_ref(transceiver);
                        g_object_set(G_OBJECT(transceiver), "direction", __convert_transceiver_direction(source->av[j].direction)->gst, NULL);
-                       source->av[j].mid = __get_gst_mid_from_transceiver(j == AV_IDX_AUDIO, GST_OBJECT_NAME(transceiver));
+                       source->av[j].mid = __get_mid_from_transceiver_count(j == AV_IDX_AUDIO, webrtc->transceiver_count++);
 
                        LOG_INFO("source->id[%u] source->av[%s][transceiver:%p, direction:%s, mid:%s]",
                                source->id, j == AV_IDX_AUDIO ? "AUDIO" : "VIDEO", source->av[j].transceiver,