Slight cleanup of media setup, leak fixes

diff --git a/chat.c b/chat.c
index 30152407b972e77baff6c481c4722bb88e950776..4a67039338b19b44d658b03cf827f419a2bb1f3e 100644 (file)
--- a/chat.c
+++ b/chat.c
@@ -283,60 +283,72 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
                                                                "fsrtpconference",
                                                                name,
                                                                TRUE);
-               if (chat->media) {
-//                     const gchar *caps = "application/x-srtp,media=(string)audio,clock-rate=(int)16000,payload=(int)97,encoding-name=(string)CHIME,stereo=(string)0";
-
-                       gboolean r = purple_media_add_stream(chat->media, "chime", name,
-                                                            PURPLE_MEDIA_AUDIO, TRUE,
-                                                            "app", 0, NULL);
-                       gchar *srcname = g_strdup_printf("chime_src_%p", call);
-                       gchar *sinkname = g_strdup_printf("chime_sink_%p", call);
-                       gchar *srcpipe = g_strdup_printf("appsrc name=%s format=time", srcname);
-                       gchar *sinkpipe = g_strdup_printf("appsink name=%s async=false", sinkname);
-
-                       PurpleMediaCandidate *cand =
-                               purple_media_candidate_new(NULL, 1,
-                                                          PURPLE_MEDIA_CANDIDATE_TYPE_HOST,
-                                                          PURPLE_MEDIA_NETWORK_PROTOCOL_UDP,
-                                                          sinkpipe, 16000);
-                       g_object_set(cand, "username", srcpipe, NULL);
-                       g_free(sinkpipe);
-                       g_free(srcpipe);
-
-                       GList *cands = g_list_append (NULL, cand);
-                       GList *codecs = g_list_append(NULL,
-                                                     purple_media_codec_new(97, "CHIME", PURPLE_MEDIA_AUDIO, 0));
-                       g_object_set(codecs->data, "channels", 1, NULL);
-
-                       purple_media_add_remote_candidates(chat->media, "chime", name, cands);
-                       if (!purple_media_set_remote_codecs(chat->media, "chime", name, codecs)) {
-                               purple_media_error(chat->media, _("Error setting Chime OPUS codec\n"));
-                               purple_media_end(chat->media, NULL, NULL);
-                               chat->media = NULL;
-                               chime_call_set_mute(chat->call, TRUE);
-                               return;
-                       }
-
-                       GstElement *pipeline = purple_media_manager_get_pipeline(mgr);
-                       GstElement *appsrc = gst_bin_get_by_name(GST_BIN(pipeline), srcname);
-                       GstElement *appsink = gst_bin_get_by_name(GST_BIN(pipeline), sinkname);
-                       g_free(srcname);
-                       g_free(sinkname);
-
-                       gst_app_src_set_size(GST_APP_SRC(appsrc), -1);
-                       gst_app_src_set_max_bytes(GST_APP_SRC(appsrc), 100);
-                       gst_app_src_set_stream_type(GST_APP_SRC(appsrc), GST_APP_STREAM_TYPE_STREAM);
-//                     gst_base_src_set_live(GST_BASE_SRC(appsrc), TRUE);
-                       chime_call_install_gst_app_callbacks(chat->call, GST_APP_SRC(appsrc), GST_APP_SINK(appsink));
-                       g_signal_connect(chat->media, "state-changed", G_CALLBACK(call_media_changed), chat);
-                       g_signal_connect(chat->media, "stream-info", G_CALLBACK(call_stream_info), chat);
-
-                       purple_media_stream_info(chat->media, PURPLE_MEDIA_INFO_ACCEPT, "chime", name, FALSE);
-                       GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(purple_media_manager_get_pipeline(mgr)), GST_DEBUG_GRAPH_SHOW_ALL, "chime conference graph");
+               if (!chat->media) {
+                       /* XX: Report error, but not with purple_media_error()! */
+                       chime_call_set_mute(chat->call, TRUE);
+                       return;
                }
-       }
 
+               if (!purple_media_add_stream(chat->media, "chime", name,
+                                            PURPLE_MEDIA_AUDIO, TRUE,
+                                            "app", 0, NULL)) {
 
+                       purple_media_error(chat->media, _("Error adding media stream\n"));
+                       purple_media_end(chat->media, NULL, NULL);
+                       chat->media = NULL;
+                       chime_call_set_mute(chat->call, TRUE);
+                       return;
+               }
+
+               gchar *srcname = g_strdup_printf("chime_src_%p", call);
+               gchar *sinkname = g_strdup_printf("chime_sink_%p", call);
+               gchar *srcpipe = g_strdup_printf("appsrc name=%s format=time", srcname);
+               gchar *sinkpipe = g_strdup_printf("appsink name=%s async=false", sinkname);
+
+               PurpleMediaCandidate *cand =
+                       purple_media_candidate_new(NULL, 1,
+                                                  PURPLE_MEDIA_CANDIDATE_TYPE_HOST,
+                                                  PURPLE_MEDIA_NETWORK_PROTOCOL_UDP,
+                                                  sinkpipe, 16000);
+               g_object_set(cand, "username", srcpipe, NULL);
+               g_free(sinkpipe);
+               g_free(srcpipe);
+
+               GList *cands = g_list_append (NULL, cand);
+               purple_media_add_remote_candidates(chat->media, "chime", name, cands);
+               purple_media_candidate_list_free(cands);
+
+               GList *codecs = g_list_append(NULL,
+                                             purple_media_codec_new(97, "CHIME", PURPLE_MEDIA_AUDIO, 0));
+               g_object_set(codecs->data, "channels", 1, NULL);
+
+               if (!purple_media_set_remote_codecs(chat->media, "chime", name, codecs)) {
+                       purple_media_codec_list_free(codecs);
+                       purple_media_error(chat->media, _("Error setting Chime OPUS codec\n"));
+                       purple_media_end(chat->media, NULL, NULL);
+                       chat->media = NULL;
+                       chime_call_set_mute(chat->call, TRUE);
+                       return;
+               }
+               purple_media_codec_list_free(codecs);
+
+               GstElement *pipeline = purple_media_manager_get_pipeline(mgr);
+               GstElement *appsrc = gst_bin_get_by_name(GST_BIN(pipeline), srcname);
+               GstElement *appsink = gst_bin_get_by_name(GST_BIN(pipeline), sinkname);
+               g_free(srcname);
+               g_free(sinkname);
+
+               gst_app_src_set_size(GST_APP_SRC(appsrc), -1);
+               gst_app_src_set_max_bytes(GST_APP_SRC(appsrc), 100);
+               gst_app_src_set_stream_type(GST_APP_SRC(appsrc), GST_APP_STREAM_TYPE_STREAM);
+//             gst_base_src_set_live(GST_BASE_SRC(appsrc), TRUE);
+               chime_call_install_gst_app_callbacks(chat->call, GST_APP_SRC(appsrc), GST_APP_SINK(appsink));
+               g_signal_connect(chat->media, "state-changed", G_CALLBACK(call_media_changed), chat);
+               g_signal_connect(chat->media, "stream-info", G_CALLBACK(call_stream_info), chat);
+
+               purple_media_stream_info(chat->media, PURPLE_MEDIA_INFO_ACCEPT, "chime", name, FALSE);
+               GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(purple_media_manager_get_pipeline(mgr)), GST_DEBUG_GRAPH_SHOW_ALL, "chime conference graph");
+       }
 }
 
 static void on_call_participants(ChimeCall *call, GHashTable *participants, struct chime_chat *chat)