static gboolean __ms_skip_set_state(media_streamer_s *ms_streamer)
{
- media_streamer_node_s *webrtc = NULL;
-
ms_retvm_if(ms_streamer == NULL, MEDIA_STREAMER_ERROR_INVALID_PARAMETER, "ms_streamer is NULL");
ms_retvm_if(ms_streamer->nodes_table == NULL, MEDIA_STREAMER_ERROR_INVALID_PARAMETER, "nodes_table is NULL");
- ms_debug_fenter();
-
- webrtc = (media_streamer_node_s *)g_hash_table_lookup(ms_streamer->nodes_table, "webrtc_container");
- if (webrtc && ms_streamer->pend_state == MEDIA_STREAMER_STATE_READY) {
- ms_debug_fleave();
+ if (g_hash_table_contains(ms_streamer->nodes_table, "webrtc_container") &&
+ ms_streamer->pend_state == MEDIA_STREAMER_STATE_READY) {
+ ms_info("Skip set state, state will be set after connecting ICE connection.");
return TRUE;
}
break;
}
- if(__ms_skip_set_state(ms_streamer)) {
- ms_info("Skip set state, state is set after connecting ICE connection.");
+ if (__ms_skip_set_state(ms_streamer)) {
g_mutex_unlock(&ms_streamer->mutex_lock);
break;
}
* set SSRC information to SDP properly inside of webrtcbin. Therefore, we set the state to PLAYING
* for a short time as below.
*/
- media_streamer_node_s *webrtc = (media_streamer_node_s *)g_hash_table_lookup(ms_streamer->nodes_table, "webrtc_container");
- if (webrtc) {
+ if (g_hash_table_contains(ms_streamer->nodes_table, "webrtc_container")) {
if (ms_pipeline_is_get_state_with_no_preroll(ms_streamer)) {
ms_info("No preroll, we make the GST state to PLAYING here");
ms_streamer->need_paused_by_live_source = TRUE;