/*
 *  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.
 */

#include "webrtc/common_video/h264/h264_common.h"

namespace webrtc {
namespace H264 {

const uint8_t kNaluTypeMask = 0x1F;

std::vector<NaluIndex> FindNaluIndices(const uint8_t* buffer,
                                       size_t buffer_size) {
  // This is sorta like Boyer-Moore, but with only the first optimization step:
  // given a 3-byte sequence we're looking at, if the 3rd byte isn't 1 or 0,
  // skip ahead to the next 3-byte sequence. 0s and 1s are relatively rare, so
  // this will skip the majority of reads/checks.
  RTC_CHECK_GE(buffer_size, kNaluShortStartSequenceSize);
  std::vector<NaluIndex> sequences;
  const size_t end = buffer_size - kNaluShortStartSequenceSize;
  for (size_t i = 0; i < end;) {
    if (buffer[i + 2] > 1) {
      i += 3;
    } else if (buffer[i + 2] == 1 && buffer[i + 1] == 0 && buffer[i] == 0) {
      // We found a start sequence, now check if it was a 3 of 4 byte one.
      NaluIndex index = {i, i + 3, 0};
      if (index.start_offset > 0 && buffer[index.start_offset - 1] == 0)
        --index.start_offset;

      // Update length of previous entry.
      auto it = sequences.rbegin();
      if (it != sequences.rend())
        it->payload_size = index.start_offset - it->payload_start_offset;

      sequences.push_back(index);

      i += 3;
    } else {
      ++i;
    }
  }

  // Update length of last entry, if any.
  auto it = sequences.rbegin();
  if (it != sequences.rend())
    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;) {
    // 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;
    } else {
      // Single rbsp byte.
      rbsp_buffer->AppendData(sps_bytes[i]);
      ++i;
    }
  }
  return rbsp_buffer;
}

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