Implement RTP keepalive in native stack.

webrtc/video/replay.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 284 matching lines @@
           fprintf(stderr, "Unknown SSRC: %u!\n", header.ssrc);
         ++unknown_packets[header.ssrc];
         break;
       }
       case PacketReceiver::DELIVERY_PACKET_ERROR: {
         fprintf(stderr, "Packet error, corrupt packets or incorrect setup?\n");
         RTPHeader header;
         std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
         parser->Parse(packet.data, packet.length, &header);
         fprintf(stderr, "Packet len=%zu pt=%u seq=%u ts=%u ssrc=0x%8x\n",
-            packet.length, header.payloadType, header.sequenceNumber,
-            header.timestamp, header.ssrc);
+                packet.length, header.payloadType, header.sequenceNumber,
+                header.timestamp, header.ssrc);
         break;
       }
     }
     if (last_time_ms != 0 && last_time_ms != packet.time_ms) {
       SleepMs(packet.time_ms - last_time_ms);
     }
     last_time_ms = packet.time_ms;
   }
   fprintf(stderr, "num_packets: %d\n", num_packets);
 
@@ skipping 10 matching lines @@
 }
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   ::testing::InitGoogleTest(&argc, argv);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   webrtc::test::RunTest(webrtc::RtpReplay);
   return 0;
 }