Allocate extra buffer space in FrameObject in case of H264.

webrtc/modules/video_coding/frame_object.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 17 matching lines @@
                                size_t frame_size,
                                int times_nacked,
                                int64_t received_time)
     : packet_buffer_(packet_buffer),
       first_seq_num_(first_seq_num),
       last_seq_num_(last_seq_num),
       received_time_(received_time),
       times_nacked_(times_nacked) {
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num);
   if (packet) {
+    // RtpFrameObject members
+    frame_type_ = packet->frameType;
+    codec_type_ = packet->codec;
+
     // TODO(philipel): Remove when encoded image is replaced by FrameObject.
     // VCMEncodedFrame members
     CopyCodecSpecific(&packet->video_header);
     _completeFrame = true;
     _payloadType = packet->payloadType;
     _timeStamp = packet->timestamp;
     ntp_time_ms_ = packet->ntp_time_ms_;
-    _buffer = new uint8_t[frame_size]();
-    _size = frame_size;
+
+    // Since FFmpeg use an optimized bitstream reader that reads in chunks of
+    // 32/64 bits we have to add at least that much padding to the buffer
+    // to make sure the decoder doesn't read out of bounds.
+    // NOTE! EncodedImage::_size is the size of the buffer (think capacity of
+    //       an std::vector) and EncodedImage::_length is the actual size of
+    //       the bitstream (think size of an std::vector).
+    if (codec_type_ == kVideoCodecH264)
+      _size = frame_size + EncodedImage::kBufferPaddingBytesH264;
+    else
+      _size = frame_size;
+
+    _buffer = new uint8_t[_size];
     _length = frame_size;
     _frameType = packet->frameType;
     GetBitstream(_buffer);
 
-    // RtpFrameObject members
-    frame_type_ = packet->frameType;
-    codec_type_ = packet->codec;
-
     // FrameObject members
     timestamp = packet->timestamp;
   }
 }
 
 RtpFrameObject::~RtpFrameObject() {
   packet_buffer_->ReturnFrame(this);
 }
 
 uint16_t RtpFrameObject::first_seq_num() const {
@@ skipping 34 matching lines @@
 
 RTPVideoTypeHeader* RtpFrameObject::GetCodecHeader() const {
   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
   if (!packet)
     return nullptr;
   return &packet->video_header.codecHeader;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc