Add sender controlled playout delay limits

webrtc/config.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.
  */
 #include "webrtc/config.h"
@@ skipping 31 matching lines @@
     "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 const int RtpExtension::kAbsSendTimeDefaultId = 3;
 
 const char* RtpExtension::kVideoRotationUri = "urn:3gpp:video-orientation";
 const int RtpExtension::kVideoRotationDefaultId = 4;
 
 const char* RtpExtension::kTransportSequenceNumberUri =
     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
+// This extension allows applications to adaptively limit the playout delay
+// on frames as per the current needs. For example, a gaming application
+// has very different needs on end-to-end delay compared to a video-conference
+// application.
+const char* RtpExtension::kPlayoutDelayUri =
+    "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
+const int RtpExtension::kPlayoutDelayDefaultId = 6;
+
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kAudioLevelUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+         uri == webrtc::RtpExtension::kPlayoutDelayUri;

[danilchap, 2016/06/03 09:01:37]

for now this extension is not used for Audio, may be remove it here.

[Irfan, 2016/06/03 15:55:33]

Done.

 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
   return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+         uri == webrtc::RtpExtension::kPlayoutDelayUri;
 }
 
 VideoStream::VideoStream()
     : width(0),
       height(0),
       max_framerate(-1),
       min_bitrate_bps(-1),
       target_bitrate_bps(-1),
       max_bitrate_bps(-1),
       max_qp(-1) {}
@@ skipping 51 matching lines @@
   }
   ss << ", encoder_specific_settings: ";
   ss << (encoder_specific_settings != NULL ? "(ptr)" : "NULL");
 
   ss << ", min_transmit_bitrate_bps: " << min_transmit_bitrate_bps;
   ss << '}';
   return ss.str();
 }
 
 }  // namespace webrtc