For #1638, #307, refactor rtc player.

trunk/research/players/rtc_player.html

@@ -53,6 +53,7 @@
     </footer>
 </div>
 <script type="text/javascript">
+    var pc = null; // Global handler to do cleanup when replaying.
     $(function(){
         $('#rtc_media_player').hide();
         var query = parse_query_string();
@@ -63,7 +64,12 @@
             var urlObject = parse_rtmp_url($("#txt_url").val());
             var schame = window.location.protocol;
 
-            var pc = new RTCPeerConnection(null);
+            // Close PC when user replay.
+            if (pc) {
+                pc.close();
+            }
+
+            pc = new RTCPeerConnection(null);
             pc.onaddstream = function (event) {
                 var player = document.getElementById('rtc_media_player');
                 player.srcObject = event.stream;

trunk/src/app/srs_app_rtp.cpp

@@ -76,9 +76,6 @@ srs_error_t SrsRtpMuxer::frame_to_packet(SrsSharedPtrMessage* shared_frame, SrsF
     for (int i = 0; i < format->video->nb_samples; ++i) {
         SrsSample sample = format->video->samples[i];
 
-        uint8_t header = sample.bytes[0];
-        uint8_t nal_type = header & kNalTypeMask;
-
         if (sample.size <= max_payload_size) {
             if ((err = packet_single_nalu(shared_frame, format, &sample, rtp_packet_vec)) != srs_success) {
                 return srs_error_wrap(err, "packet single nalu");