New jitter buffer experiment.

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 201 matching lines @@
           break;
 
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;
         start_seq_num--;
       }
 
       std::unique_ptr<RtpFrameObject> frame(
           new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
                              max_nack_count, clock_->TimeInMilliseconds()));
 
+      crit_.Leave();
       received_frame_callback_->OnReceivedFrame(std::move(frame));
+      crit_.Enter();

[stefan-webrtc, 2016/11/08 10:41:34]

I think we should rewrite the code so that we can properly use scoped crits. For
instance you can return a list of frames to be sent out from this method and
call the callback from InsertPacket() after you have left the crit.

[philipel, 2016/11/08 12:28:39]

Done.

     }
 
     ++seq_num;
   }
 }
 
 void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
   rtc::CritScope lock(&crit_);
   size_t index = frame->first_seq_num() % size_;
   size_t end = (frame->last_seq_num() + 1) % size_;
@@ skipping 27 matching lines @@
     size_t length = data_buffer_[index].sizeBytes;
     memcpy(destination, source, length);
     destination += length;
     index = (index + 1) % size_;
     ++seq_num;
   }
   return true;
 }
 
 VCMPacket* PacketBuffer::GetPacket(uint16_t seq_num) {
-  rtc::CritScope lock(&crit_);
   size_t index = seq_num % size_;
   if (!sequence_buffer_[index].used ||
       seq_num != sequence_buffer_[index].seq_num) {
     return nullptr;
   }
   return &data_buffer_[index];
 }
 
 int PacketBuffer::AddRef() const {
   return rtc::AtomicOps::Increment(&ref_count_);
 }
 
 int PacketBuffer::Release() const {
   int count = rtc::AtomicOps::Decrement(&ref_count_);
   if (!count) {
     delete this;
   }
   return count;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc