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

webrtc/modules/video_coding/packet_buffer.h

 /*
  *  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 46 matching lines @@
   void Clear();
   void PaddingReceived(uint16_t seq_num);
 
   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
   rtc::Optional<int64_t> LastReceivedPacketMs() const;
   rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
 
   int AddRef() const;
   int Release() const;
 
+  size_t size_ GUARDED_BY(crit_);
+
  protected:
   // Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
   PacketBuffer(Clock* clock,
                size_t start_buffer_size,
                size_t max_buffer_size,
                OnReceivedFrameCallback* frame_callback);
 
  private:
   friend RtpFrameObject;
   // Since we want the packet buffer to be as packet type agnostic
@@ skipping 44 matching lines @@
 
   // Mark all slots used by |frame| as not used.
   // Virtual for testing.
   virtual void ReturnFrame(RtpFrameObject* frame);
 
   void UpdateMissingPackets(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   rtc::CriticalSection crit_;
 
   // Buffer size_ and max_size_ must always be a power of two.
-  size_t size_ GUARDED_BY(crit_);
   const size_t max_size_;
 
   // The fist sequence number currently in the buffer.
   uint16_t first_seq_num_ GUARDED_BY(crit_);
 
   // If the packet buffer has received its first packet.
   bool first_packet_received_ GUARDED_BY(crit_);
 
   // If the buffer is cleared to |first_seq_num_|.
   bool is_cleared_to_first_seq_num_ GUARDED_BY(crit_);
@@ skipping 16 matching lines @@
   std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
       GUARDED_BY(crit_);
 
   mutable volatile int ref_count_ = 0;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_