Piggybacking simulcast id and ALR experiment id into video content type extension.

webrtc/common_video/include/video_frame.h

 /*
  *  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 36 matching lines @@
   uint32_t _encodedHeight = 0;
   uint32_t _timeStamp = 0;
   // NTP time of the capture time in local timebase in milliseconds.
   int64_t ntp_time_ms_ = 0;
   int64_t capture_time_ms_ = 0;
   FrameType _frameType = kVideoFrameDelta;
   uint8_t* _buffer;
   size_t _length;
   size_t _size;
   VideoRotation rotation_ = kVideoRotation_0;
-  VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
+  mutable VideoContentType content_type_ = VideoContentType::UNSPECIFIED;

[sprang_webrtc, 2017/08/24 09:13:16]

Why does it need to be mutable?

[ilnik, 2017/08/25 12:35:07]

Because it's modified in the VCMEncodedFrameCallback, yet frame there is a const
reference.

   bool _completeFrame = false;
   AdaptReason adapt_reason_;
   int qp_ = -1;  // Quantizer value.
 
   // When an application indicates non-zero values here, it is taken as an
   // indication that all future frames will be constrained with those limits
   // until the application indicates a change again.
   PlayoutDelay playout_delay_ = {-1, -1};
 
   // Timing information should be updatable on const instances.
   mutable struct Timing {
     uint8_t flags = TimingFrameFlags::kInvalid;
     int64_t encode_start_ms = 0;
     int64_t encode_finish_ms = 0;
     int64_t packetization_finish_ms = 0;
     int64_t pacer_exit_ms = 0;
     int64_t network_timestamp_ms = 0;
     int64_t network2_timestamp_ms = 0;
     int64_t receive_start_ms = 0;
     int64_t receive_finish_ms = 0;
   } timing_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_