[EXPERIMENTAL] Generic stereo codec with index header sending merged frames

webrtc/modules/video_coding/packet_buffer.cc

 /*
  *  Copyright (c) 2016 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 19 matching lines @@
     size_t max_buffer_size,
     OnReceivedFrameCallback* received_frame_callback) {
   return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer(
       clock, start_buffer_size, max_buffer_size, received_frame_callback));
 }
 
 PacketBuffer::PacketBuffer(Clock* clock,
                            size_t start_buffer_size,
                            size_t max_buffer_size,
                            OnReceivedFrameCallback* received_frame_callback)
-    : clock_(clock),
-      size_(start_buffer_size),
+    : size_(start_buffer_size),
+      clock_(clock),
       max_size_(max_buffer_size),
       first_seq_num_(0),
       first_packet_received_(false),
       is_cleared_to_first_seq_num_(false),
       data_buffer_(start_buffer_size),
       sequence_buffer_(start_buffer_size),
       received_frame_callback_(received_frame_callback) {
   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
   // Buffer size must always be a power of 2.
   RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
@@ skipping 350 matching lines @@
       missing_packets_.insert(*newest_inserted_seq_num_);
       ++*newest_inserted_seq_num_;
     }
   } else {
     missing_packets_.erase(seq_num);
   }
 }
 
 }  // namespace video_coding
 }  // namespace webrtc