Break backwards traversal loop if we have looped around all packet in the PacketBuffer for H264 fra…

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 179 matching lines @@
   }
   if (sequence_buffer_[prev_index].continuous)
     return true;
 
   return false;
 }
 
 std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
     uint16_t seq_num) {
   std::vector<std::unique_ptr<RtpFrameObject>> found_frames;
-  size_t packets_tested = 0;
-  while (packets_tested < size_ && PotentialNewFrame(seq_num)) {
+  for (size_t i = 0; i < size_ && PotentialNewFrame(seq_num); ++i) {
     size_t index = seq_num % size_;
     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) {
       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) {
+
+      // Since packet at |data_buffer_[index]| is already part of the frame
+      // we will have at most |size_ - 1| packets left to check.
+      for (size_t j = 0; j < size_ - 1; ++j) {
         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 (!is_h264 && sequence_buffer_[start_index].frame_begin)
           break;
 
         start_index = start_index > 0 ? start_index - 1 : size_ - 1;
 
         // 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) &&
+        if (is_h264 &&
             (!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;
 }
 
 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_;
   uint16_t seq_num = frame->first_seq_num();
   while (index != end) {
@@ skipping 47 matching lines @@
 int PacketBuffer::Release() const {
   int count = rtc::AtomicOps::Decrement(&ref_count_);
   if (!count) {
     delete this;
   }
   return count;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc