Optimize ParseRbsp method in H264 bitstream parser.

webrtc/common_video/h264/h264_common.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 43 matching lines @@
     it->payload_size = buffer_size - it->payload_start_offset;
 
   return sequences;
 }
 
 NaluType ParseNaluType(uint8_t data) {
   return static_cast<NaluType>(data & kNaluTypeMask);
 }
 
 std::unique_ptr<rtc::Buffer> ParseRbsp(const uint8_t* data, size_t length) {
-  std::unique_ptr<rtc::Buffer> rbsp_buffer(new rtc::Buffer(0, length));
-  const char* sps_bytes = reinterpret_cast<const char*>(data);
-  for (size_t i = 0; i < length;) {
+  std::unique_ptr<uint8_t[]> out(new uint8_t[length]);

[magjed_webrtc, 2017/03/03 13:14:14]

I think it would be better with an std::vector<uint8_t> here, and use push_back
instead of |j|. Also, we can change the return type to std::vector and that
removes an extra copy.

+
+  size_t j, i;
+  for (i = j = 0; i < length;) {
+    uint8_t* dst = out.get();
     // Be careful about over/underflow here. byte_length_ - 3 can underflow, and
     // i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_
     // above, and that expression will produce the number of bytes left in
     // the stream including the byte at i.
-    if (length - i >= 3 && data[i] == 0 && data[i + 1] == 0 &&
-        data[i + 2] == 3) {
-      // Two rbsp bytes + the emulation byte.
-      rbsp_buffer->AppendData(sps_bytes + i, 2);
-      i += 3;
+    if (length - i >= 3 && !data[i] && !data[i + 1] && data[i + 2] == 3) {
+      // Two rbsp bytes.
+      dst[j++] = data[i++];
+      dst[j++] = data[i++];
+      // Skip the emulation byte.
+      i++;
     } else {
       // Single rbsp byte.
-      rbsp_buffer->AppendData(sps_bytes[i]);
-      ++i;
+      dst[j++] = data[i++];
     }
   }
-  return rbsp_buffer;
+  return std::unique_ptr<rtc::Buffer>(new rtc::Buffer(out.get(), j));
 }
 
 void WriteRbsp(const uint8_t* bytes, size_t length, rtc::Buffer* destination) {
   static const uint8_t kZerosInStartSequence = 2;
   static const uint8_t kEmulationByte = 0x03u;
   size_t num_consecutive_zeros = 0;
   destination->EnsureCapacity(destination->size() + length);
 
   for (size_t i = 0; i < length; ++i) {
     uint8_t byte = bytes[i];
     if (byte <= kEmulationByte &&
         num_consecutive_zeros >= kZerosInStartSequence) {
       // Need to escape.
       destination->AppendData(kEmulationByte);
       num_consecutive_zeros = 0;
     }
     destination->AppendData(byte);
     if (byte == 0) {
       ++num_consecutive_zeros;
     } else {
       num_consecutive_zeros = 0;
     }
   }
 }
 
 }  // namespace H264
 }  // namespace webrtc