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 58 matching lines @@
 
 int PacketBuffer::InsertPacket(Packet* packet) {
   if (!packet || packet->empty()) {
     if (packet) {
       delete packet;
     }
     LOG(LS_WARNING) << "InsertPacket invalid packet";
     return kInvalidPacket;
   }
 
+  RTC_DCHECK_GE(packet->priority.codec_level, 0);
+  RTC_DCHECK_GE(packet->priority.red_level, 0);

[kwiberg-webrtc, 2016/09/20 14:56:23]

...or maybe make the invariant-checking function public...

[ossu, 2016/09/20 15:45:21]

Acknowledged.

[ossu, 2016/09/21 10:23:13]

Considered it but decided to keep the check like this here, so that we get a
better error message when this happens, i.e. it'll point to one of these lines,
rather than inside Priority::CheckInvariant(). I bet that if priority levels are
set wrong, that will be caught upon insertion into the PacketBuffer (since
they're ferried more-or-less directly from ParsePayload to the PacketBuffer).

[kwiberg-webrtc, 2016/09/21 10:59:13]

Acknowledged.

+
   int return_val = kOK;
 
   packet->waiting_time = tick_timer_->GetNewStopwatch();
 
   if (buffer_.size() >= max_number_of_packets_) {
     // Buffer is full. Flush it.
     Flush();
     LOG(LS_WARNING) << "Packet buffer flushed";
     return_val = kFlushed;
   }
@@ skipping 166 matching lines @@
   }
   return 0;
 }
 
 int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
   return DiscardOldPackets(timestamp_limit, 0);
 }
 
 void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type) {
   for (auto it = buffer_.begin(); it != buffer_.end(); /* */) {
-    Packet *packet = *it;
+    Packet* packet = *it;
     if (packet->header.payloadType == payload_type) {
       delete packet;
       it = buffer_.erase(it);
     } else {
       ++it;
     }
   }
 }
 
 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(hlundin): Verify that it's fine to count all packets and remove
+      // this check.
+      if (packet->priority != Packet::Priority(0, 0)) {
         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