video_coding::PacketBuffer now group all H264 packets with the same timestamp into the same frame.

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 193 matching lines @@
     // 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) {
       size_t frame_size = 0;
       int max_nack_count = -1;
       uint16_t start_seq_num = seq_num;
 
       // Find the start index by searching backward until the packet with
       // the |frame_begin| flag is set.
       int start_index = index;
+
+      bool is_h264 = data_buffer_[start_index].codec == kVideoCodecH264;
+      int64_t frame_timestamp = data_buffer_[start_index].timestamp;
       while (true) {
         frame_size += data_buffer_[start_index].sizeBytes;
         max_nack_count =
             std::max(max_nack_count, data_buffer_[start_index].timesNacked);
         sequence_buffer_[start_index].frame_created = true;
 
-        if (sequence_buffer_[start_index].frame_begin)
+        if (!is_h264 && sequence_buffer_[start_index].frame_begin)
           break;
 
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;
-        start_seq_num--;
+
+        // In the case of H264 we don't have a frame_begin bit (yes,
+        // |frame_begin| might be set to true but that is a lie). So instead
+        // we traverese backwards as long as we have a previous packet and
+        // the timestamp of that packet is the same as this one. This may cause
+        // the PacketBuffer to hand out incomplete frames.
+        // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106
+        //
+        // Since we ignore the |frame_begin| flag of the inserted packets
+        // we check that |start_index != static_cast<int>(index)| to make sure
+        // that we don't get stuck in a loop if the packet buffer is filled
+        // with packets of the same timestamp.
+        if (is_h264 && start_index != static_cast<int>(index) &&
+            (!sequence_buffer_[start_index].used ||
+             data_buffer_[start_index].timestamp != frame_timestamp)) {
+          break;
+        }
+
+        --start_seq_num;
       }
 
       found_frames.emplace_back(
           new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
                              max_nack_count, clock_->TimeInMilliseconds()));
     }
     ++seq_num;
     ++packets_tested;
   }
   return found_frames;
@@ skipping 55 matching lines @@
 int PacketBuffer::Release() const {
   int count = rtc::AtomicOps::Decrement(&ref_count_);
   if (!count) {
     delete this;
   }
   return count;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc