Ignore aud and sei NALus when parsing bitstream.

webrtc/common_video/h264/h264_bitstream_parser.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include "webrtc/common_video/h264/h264_bitstream_parser.h"
@@ skipping 266 matching lines @@
         LOG(LS_WARNING) << "Unable to parse SPS from H264 bitstream.";
       break;
     }
     case H264::NaluType::kPps: {
       pps_ = PpsParser::ParsePps(slice + H264::kNaluTypeSize,
                                  length - H264::kNaluTypeSize);
       if (!pps_)
         LOG(LS_WARNING) << "Unable to parse PPS from H264 bitstream.";
       break;
     }
+    case H264::NaluType::kAud:
+    case H264::NaluType::kSei:
+      break;  // Ignore these nalus, as we don't care about their contents.
     default:
       Result res = ParseNonParameterSetNalu(slice, length, nalu_type);
       if (res != kOk)
         LOG(LS_INFO) << "Failed to parse bitstream. Error: " << res;
       break;
   }
 }
 
 void H264BitstreamParser::ParseBitstream(const uint8_t* bitstream,
                                          size_t length) {
   std::vector<H264::NaluIndex> nalu_indices =
       H264::FindNaluIndices(bitstream, length);
   for (const H264::NaluIndex& index : nalu_indices)
     ParseSlice(&bitstream[index.payload_start_offset], index.payload_size);
 }
 
 bool H264BitstreamParser::GetLastSliceQp(int* qp) const {
   if (!last_slice_qp_delta_ || !pps_)
     return false;
   const int parsed_qp = 26 + pps_->pic_init_qp_minus26 + *last_slice_qp_delta_;
   if (parsed_qp < kMinQpValue || parsed_qp > kMaxQpValue) {
     LOG(LS_ERROR) << "Parsed invalid QP from bitstream.";
     return false;
   }
   *qp = parsed_qp;
   return true;
 }
 
 }  // namespace webrtc