PacketBuffer now can save how many times a packet has been nacked.

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 19 matching lines @@
       first_packet_received_(false),
       data_buffer_(start_buffer_size),
       sequence_buffer_(start_buffer_size),
       reference_finder_(frame_callback) {
   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
   // Buffer size must always be a power of 2.
   RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
   RTC_DCHECK((max_buffer_size & (max_buffer_size - 1)) == 0);
 }
 
-bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
+bool PacketBuffer::InsertPacket(const VCMPacket& packet, int times_nacked) {
   rtc::CritScope lock(&crit_);
   uint16_t seq_num = packet.seqNum;
   size_t index = seq_num % size_;
 
   if (!first_packet_received_) {
     first_seq_num_ = seq_num - 1;
     last_seq_num_ = seq_num;
     first_packet_received_ = true;
   }
 
@@ skipping 14 matching lines @@
 
   if (AheadOf(seq_num, last_seq_num_))
     last_seq_num_ = seq_num;
 
   sequence_buffer_[index].frame_begin = packet.isFirstPacket;
   sequence_buffer_[index].frame_end = packet.markerBit;
   sequence_buffer_[index].seq_num = packet.seqNum;
   sequence_buffer_[index].continuous = false;
   sequence_buffer_[index].frame_created = false;
   sequence_buffer_[index].used = true;
+  sequence_buffer_[index].times_nacked = times_nacked;
   data_buffer_[index] = packet;
 
   FindFrames(seq_num);
   return true;
 }
 
 void PacketBuffer::ClearTo(uint16_t seq_num) {
   rtc::CritScope lock(&crit_);
   size_t index = first_seq_num_ % size_;
   while (AheadOf<uint16_t>(seq_num, first_seq_num_ + 1)) {
@@ skipping 46 matching lines @@
 
 void PacketBuffer::FindFrames(uint16_t seq_num) {
   size_t index = seq_num % size_;
   while (IsContinuous(seq_num)) {
     sequence_buffer_[index].continuous = true;
 
     // If all packets of the frame is continuous, find the first packet of the
     // frame and create an RtpFrameObject.
     if (sequence_buffer_[index].frame_end) {
       int start_index = index;
+      size_t combined_size = 0;

[stefan-webrtc, 2016/05/20 13:10:23]

Should this be frame_size instead?

[philipel, 2016/05/23 09:19:22]

Done.

+      int8_t max_nack_count = -1;
       uint16_t start_seq_num = seq_num;
 
       while (!sequence_buffer_[start_index].frame_begin) {
+        combined_size += data_buffer_[start_index].sizeBytes;
+        max_nack_count = std::max<int8_t>(
+            max_nack_count, sequence_buffer_[start_index].times_nacked);
         sequence_buffer_[start_index].frame_created = true;
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;

[stefan-webrtc, 2016/05/20 13:10:23]

start_index seems like a bad variable name as it doesn't point to the start all
the time.

[philipel, 2016/05/23 09:19:22]

Hard to find a better name, I added a comment about it.

         start_seq_num--;
       }
+      combined_size += data_buffer_[start_index].sizeBytes;
+      max_nack_count = std::max<int8_t>(
+          max_nack_count, sequence_buffer_[start_index].times_nacked);
       sequence_buffer_[start_index].frame_created = true;
 
-      std::unique_ptr<RtpFrameObject> frame(
-          new RtpFrameObject(this, start_seq_num, seq_num));
+      std::unique_ptr<RtpFrameObject> frame(new RtpFrameObject(
+          this, start_seq_num, seq_num, combined_size, max_nack_count));
       reference_finder_.ManageFrame(std::move(frame));
     }
 
     index = (index + 1) % size_;
     ++seq_num;
   }
 }
 
 void PacketBuffer::ReturnFrame(RtpFrameObject* frame) {
   rtc::CritScope lock(&crit_);
@@ skipping 52 matching lines @@
 void PacketBuffer::Clear() {
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < size_; ++i)
     sequence_buffer_[i].used = false;
 
   first_packet_received_ = false;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc