New jitter buffer experiment.

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 82 matching lines @@
 
   // Tries to expand the buffer.
   bool ExpandBufferSize() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   // Test if all previous packets has arrived for the given sequence number.
   bool PotentialNewFrame(uint16_t seq_num) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   // Test if all packets of a frame has arrived, and if so, creates a frame.
   // May create multiple frames per invocation.
-  void FindFrames(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void FindFrames(uint16_t seq_num);
 
   // Copy the bitstream for |frame| to |destination|.
   // Virtual for testing.
   virtual bool GetBitstream(const RtpFrameObject& frame, uint8_t* destination);
 
   // Get the packet with sequence number |seq_num|.
   // Virtual for testing.
-  virtual VCMPacket* GetPacket(uint16_t seq_num);
+  virtual VCMPacket* GetPacket(uint16_t seq_num)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   // Mark all slots used by |frame| as not used.
   // Virtual for testing.
   virtual void ReturnFrame(RtpFrameObject* frame);
 
   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_;
@@ skipping 20 matching lines @@
   // Called when a received frame is found.
   OnReceivedFrameCallback* const received_frame_callback_;
 
   mutable volatile int ref_count_ = 0;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_