Add sender controlled playout delay limits

webrtc/common_types.h

 /*
  *  Copyright (c) 2012 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 733 matching lines @@
   }
 
   int64_t timestamp;   // Receive time after socket delivers the data.
   int64_t not_before;  // Earliest possible time the data could have arrived,
                        // indicating the potential error in the |timestamp|
                        // value,in case the system is busy.
                        // For example, the time of the last select() call.
                        // If unknown, this value will be set to zero.
 };
 
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// A value < 0 indicates no change from previous valid value.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+struct PlayoutDelay {

[danilchap, 2016/05/26 09:41:58]

May be you want to add a default constructor and initialize values to -1/-1
Currently explicit default constructor of RTPHeaderExtension leaves this two
values uninitialized.

[Sergey Ulanov, 2016/05/30 07:22:56]

Or just add default initalizers '= -1' in the field declarations.

[Irfan, 2016/06/01 08:38:33]

I have added a comment here, but this eventually ends up in a union that gets
created and used in places.  We have to let the owner control initialization.

http://www.stroustrup.com/C++11FAQ.html#unions

I think RTP header structures deserve a clean up moving away from a union.

+  int min_ms;
+  int max_ms;
+};
+
 struct RTPHeaderExtension {
   RTPHeaderExtension();
 
   bool hasTransmissionTimeOffset;
   int32_t transmissionTimeOffset;
   bool hasAbsoluteSendTime;
   uint32_t absoluteSendTime;
   bool hasTransportSequenceNumber;
   uint16_t transportSequenceNumber;
 
   // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
   // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
   bool hasAudioLevel;
   bool voiceActivity;
   uint8_t audioLevel;
 
   // For Coordination of Video Orientation. See
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
   // ts_126114v120700p.pdf
   bool hasVideoRotation;
   uint8_t videoRotation;
+
+  PlayoutDelay playout_delay;
 };
 
 struct RTPHeader {
   RTPHeader();
 
   bool markerBit;
   uint8_t payloadType;
   uint16_t sequenceNumber;
   uint32_t timestamp;
   uint32_t ssrc;
@@ skipping 108 matching lines @@
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 enum NetworkState {
   kNetworkUp,
   kNetworkDown,
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_