Moved Opus-specific payload splitting into AudioDecoderOpus.

webrtc/modules/audio_coding/neteq/packet_buffer.cc

 /*
  *  Copyright (c) 2012 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 251 matching lines @@
 
 size_t PacketBuffer::NumPacketsInBuffer() const {
   return buffer_.size();
 }
 
 size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
   size_t num_samples = 0;
   size_t last_duration = last_decoded_length;
   for (Packet* packet : buffer_) {
     if (packet->frame) {
-      if (!packet->primary) {
+      // TODO(ossu): Check that this is really want we want to filter on.

[hlundin-webrtc, 2016/09/15 09:33:30]

I would argue that also the old code was wrong. This is a legacy from when
duplicates were purged on extracting packets (i.e., as part of GetAudio). So,
when checking the size of the buffer, we wanted to avoid counting two packets
covering the same samples twice.

Since some time back, we're purging duplicates on InsertPacket. So, the right
thing to do should be to count all packets. You can put a TODO on me for that
here. It's probably going to mess up unit tests. And, keep the new code for now.

[ossu, 2016/09/15 12:22:53]

Acknowledged.

[ossu, 2016/09/19 14:07:34]

I tried putting something else here: checking only the red_level or removing the
check altogether, but the Opus bitexactness test failed. Everything else seemed
fine, though. I've kept the behavior, but gotten rid of kHighestPriority while
we figure out what to do with it.

+      if (packet->priority != Packet::kHighestPriority) {
         continue;
       }
       size_t duration = packet->frame->Duration();
       if (duration > 0) {
         last_duration = duration;  // Save the most up-to-date (valid) duration.
       }
     }
     num_samples += last_duration;
   }
   return num_samples;
@@ skipping 14 matching lines @@
     // Continue while the list is not empty.
   }
 }
 
 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
   *num_packets = static_cast<int>(buffer_.size());
   *max_num_packets = static_cast<int>(max_number_of_packets_);
 }
 
 }  // namespace webrtc