Improve pad-added related codes of WebRTC node

- Add linking pad added by decodebin to webrtc_container's ghost_pad(video_out)
- Fix adding decodebin to webrtc_container

[Version] 0.1.60
[Issue Type] Improvement

Change-Id: I5a9fce745e78c06561fab4c77baf5aee0bf5f81c
Signed-off-by: Hyunil <hyunil46.park@samsung.com>

diff --git a/packaging/capi-media-streamer.spec b/packaging/capi-media-streamer.spec
index 7a4f903..76bb9ef 100644 (file)
--- a/packaging/capi-media-streamer.spec
+++ b/packaging/capi-media-streamer.spec
@@ -1,6 +1,6 @@
 Name:       capi-media-streamer
 Summary:    A Media Streamer API
-Version:    0.1.59
+Version:    0.1.60
 Release:    0
 Group:      Multimedia/API
 License:    Apache-2.0

diff --git a/src/media_streamer_gst_webrtc.c b/src/media_streamer_gst_webrtc.c
index 0bc478e..dc354de 100644 (file)
--- a/src/media_streamer_gst_webrtc.c
+++ b/src/media_streamer_gst_webrtc.c
@@ -51,7 +51,7 @@ int ms_webrtcbin_set_pad_format(GstElement *webrtc_container, const char *pad_na
                g_strrstr(pad_name, MS_RTP_PAD_VIDEO_IN)) {
                media = "video";
                payload = 96;
-       } else if (!media_format_get_audio_info(fmt, &mime, NULL, NULL, NULL, NULL)     &&
+       } else if (!media_format_get_audio_info(fmt, &mime, NULL, NULL, NULL, NULL) &&
                g_strrstr(pad_name, MS_RTP_PAD_AUDIO_IN)) {
                media = "audio";
                payload = 97;
@@ -545,6 +545,8 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
        media_streamer_node_s *webrtc_node = (media_streamer_node_s *)user_data;
        const gchar *new_pad_name;
        const gchar *media_type;
+       GstPad *src_pad = NULL;
+       const gchar *src_pad_name = NULL;
 
        ms_retm_if(decodebin == NULL, "decodebin is NULL");
        ms_retm_if(new_pad == NULL, "new_pad is NULL");
@@ -552,20 +554,43 @@ static void __decodebin_pad_added_cb(GstElement *decodebin, GstPad *new_pad, gpo
        ms_retm_if(GST_PAD_DIRECTION(new_pad) != GST_PAD_SRC, "new_pad is not for source");
        ms_retm_if(gst_pad_has_current_caps(new_pad) == FALSE, "new_pad does not have caps");
 
-       new_pad_name = GST_PAD_NAME(new_pad);
        media_type = gst_structure_get_name(gst_caps_get_structure(gst_pad_get_current_caps(new_pad), 0));
+       ms_debug("type is [%s]", media_type);
+
+       if (MS_ELEMENT_IS_VIDEO(media_type)) {
+               src_pad_name = MS_RTP_PAD_VIDEO_OUT;
+       } else if (MS_ELEMENT_IS_AUDIO(media_type)) {
+               src_pad_name = MS_RTP_PAD_AUDIO_OUT;
+       } else {
+               ms_error("Not supported media type[%s]", media_type);
+               return;
+       }
+
+       src_pad = gst_element_get_static_pad(webrtc_node->gst_element, src_pad_name);
+       if (!src_pad) {
+               ms_error("Failed to get pad of %s", src_pad_name);
+               return;
+       }
+       new_pad_name = GST_PAD_NAME(src_pad);
+
+       if (!gst_ghost_pad_set_target(GST_GHOST_PAD(src_pad), new_pad)) {
+               ms_error("Failed to link %s:%s", src_pad_name, GST_PAD_NAME(new_pad));
+               return;
+       }
 
        ms_debug("new_pad_name[%s], media_type[%s]", new_pad_name, media_type);
 
        __trigger_decoded_ready_callback(webrtc_node, new_pad_name, media_type);
+
+       gst_object_unref(src_pad);
 }
 
 void ms_webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpointer user_data)
 {
        media_streamer_node_s *webrtc_node = (media_streamer_node_s *)user_data;
        media_streamer_webrtc_callbacks_s *_callbacks;
-       GstPad *sink_pad;
-       GstElement *decodebin;
+       GstPad *sink_pad = NULL;
+       GstElement *decodebin = NULL;
 
        ms_retm_if(new_pad == NULL, "new_pad is NULL");
        ms_retm_if(webrtc_node == NULL, "webrtc_node is NULL");
@@ -585,18 +610,17 @@ void ms_webrtcbin_pad_added_cb(GstElement *webrtcbin, GstPad *new_pad, gpointer
        decodebin = ms_element_create(DEFAULT_DECODEBIN, NULL);
        ms_retm_if(decodebin == NULL, "decodebin is NULL");
 
-       ms_retm_if(webrtc_node->parent_streamer == NULL, "parent_streamer is NULL");
-       ms_retm_if(webrtc_node->parent_streamer->pipeline == NULL, "pipeline is NULL");
-       gst_bin_add(GST_BIN(webrtc_node->parent_streamer->pipeline), decodebin);
-
+       ms_retm_if(webrtc_node->gst_element == NULL, "webrtc_container is NULL");
+       gst_bin_add(GST_BIN(webrtc_node->gst_element), decodebin);
        gst_element_sync_state_with_parent(decodebin);
 
        g_signal_connect(decodebin, "pad-added", G_CALLBACK(__decodebin_pad_added_cb), webrtc_node);
-
        sink_pad = gst_element_get_static_pad(decodebin, "sink");
        ms_retm_if(sink_pad == NULL, "sink_pad is NULL");
 
-       gst_pad_link(new_pad, sink_pad);
+       if (gst_pad_link(new_pad, sink_pad) != GST_PAD_LINK_OK)
+               ms_error("Failed to link %s:%s", GST_PAD_NAME(new_pad), GST_PAD_NAME(sink_pad));
+
        gst_object_unref(sink_pad);
 
        ms_debug_fleave();
@@ -614,6 +638,9 @@ GstElement *ms_webrtc_element_create(void)
        ms_retvm_if(!webrtc_container, (GstElement *) NULL, "Failed to create webrtc container");
 
        ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_VIDEO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_VIDEO_OUT, GST_PAD_SRC);
+       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_AUDIO_IN, GST_PAD_SINK);
+       ms_add_no_target_ghostpad(webrtc_container, MS_RTP_PAD_AUDIO_OUT, GST_PAD_SRC);
 
        MS_SET_INT_STATIC_STRING_PARAM(webrtc_container, MEDIA_STREAMER_PARAM_WEBRTC_PEER_TYPE, DEFAULT_WEBRTC_PEER);
        MS_SET_INT_STATIC_STRING_PARAM(webrtc_container, MEDIA_STREAMER_PARAM_WEBRTC_STUN_SERVER, DEFAULT_WEBRTC_STUN_SERVER);