From: Sangchul Lee <sc11.lee@samsung.com>
Date: Tue, 9 Jan 2024 03:57:39 +0000 (+0900)
Subject: webrtc_transceiver: Get mid from transceiver name
X-Git-Tag: accepted/tizen/8.0/unified/20240112.155130~3
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=8d3b6122fa291248d205edf7eb122a39c7fd96c5;p=platform%2Fcore%2Fapi%2Fwebrtc.git

webrtc_transceiver: Get mid from transceiver name

This mid value will be set later to rtp header extension
for simulcast preparation.

[Version] 0.4.31
[Issue Type] Improvement

Change-Id: Ie85ddca278eb6a179e1753e5618f010a140ef8e2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
---

diff --git a/include/webrtc_private.h b/include/webrtc_private.h
index 1b49dc96..31299052 100644
--- a/include/webrtc_private.h
+++ b/include/webrtc_private.h
@@ -572,6 +572,7 @@ typedef struct _webrtc_gst_slot_s {
 	struct {
 		GstWebRTCRTPTransceiver *transceiver;
 		webrtc_transceiver_direction_e direction;
+		gchar *mid;
 		const char *codec;
 		GstPad *src_pad;
 		gulong src_pad_probe_id;
diff --git a/packaging/capi-media-webrtc.spec b/packaging/capi-media-webrtc.spec
index 11c130ef..40a9658b 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:    0.4.30
+Version:    0.4.31
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0
diff --git a/src/webrtc_source.c b/src/webrtc_source.c
index f6aafa79..a6d6843b 100644
--- a/src/webrtc_source.c
+++ b/src/webrtc_source.c
@@ -556,6 +556,7 @@ void _source_slot_destroy_cb(gpointer data)
 
 		if (source->av[i].transceiver)
 			gst_object_unref(source->av[i].transceiver);
+		g_free(source->av[i].mid);
 	}
 
 	if (source->bin) {
diff --git a/src/webrtc_transceiver.c b/src/webrtc_transceiver.c
index 35f904a1..afaf10c7 100644
--- a/src/webrtc_transceiver.c
+++ b/src/webrtc_transceiver.c
@@ -75,6 +75,15 @@ 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)
+{
+	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]);
+}
 
 void _webrtcbin_on_new_transceiver_cb(GstElement *webrtcbin, GstWebRTCRTPTransceiver *transceiver, gpointer user_data)
 {
@@ -124,10 +133,11 @@ 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));
 
-			LOG_INFO("source->id[%u] transceiver[%p for %s, direction:%s]",
-				source->id, source->av[j].transceiver, j == AV_IDX_AUDIO ? "AUDIO" : "VIDEO",
-				__convert_transceiver_direction(source->av[j].direction)->str);
+			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,
+				__convert_transceiver_direction(source->av[j].direction)->str, source->av[j].mid);
 			return;
 		}
 	}