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/*
* 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