108 lines
3.5 KiB
C++
108 lines
3.5 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_video/h264/h264_common.h"
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namespace webrtc {
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namespace H264 {
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const uint8_t kNaluTypeMask = 0x1F;
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std::vector<NaluIndex> FindNaluIndices(const uint8_t* buffer,
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size_t buffer_size) {
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// This is sorta like Boyer-Moore, but with only the first optimization step:
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// given a 3-byte sequence we're looking at, if the 3rd byte isn't 1 or 0,
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// skip ahead to the next 3-byte sequence. 0s and 1s are relatively rare, so
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// this will skip the majority of reads/checks.
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RTC_CHECK_GE(buffer_size, kNaluShortStartSequenceSize);
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std::vector<NaluIndex> sequences;
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const size_t end = buffer_size - kNaluShortStartSequenceSize;
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for (size_t i = 0; i < end;) {
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if (buffer[i + 2] > 1) {
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i += 3;
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} else if (buffer[i + 2] == 1 && buffer[i + 1] == 0 && buffer[i] == 0) {
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// We found a start sequence, now check if it was a 3 of 4 byte one.
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NaluIndex index = {i, i + 3, 0};
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if (index.start_offset > 0 && buffer[index.start_offset - 1] == 0)
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--index.start_offset;
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// Update length of previous entry.
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auto it = sequences.rbegin();
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if (it != sequences.rend())
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it->payload_size = index.start_offset - it->payload_start_offset;
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sequences.push_back(index);
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i += 3;
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} else {
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++i;
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}
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}
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// Update length of last entry, if any.
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auto it = sequences.rbegin();
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if (it != sequences.rend())
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it->payload_size = buffer_size - it->payload_start_offset;
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return sequences;
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}
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NaluType ParseNaluType(uint8_t data) {
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return static_cast<NaluType>(data & kNaluTypeMask);
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}
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std::unique_ptr<rtc::Buffer> ParseRbsp(const uint8_t* data, size_t length) {
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std::unique_ptr<rtc::Buffer> rbsp_buffer(new rtc::Buffer(0, length));
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const char* sps_bytes = reinterpret_cast<const char*>(data);
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for (size_t i = 0; i < length;) {
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// Be careful about over/underflow here. byte_length_ - 3 can underflow, and
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// i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_
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// above, and that expression will produce the number of bytes left in
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// the stream including the byte at i.
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if (length - i >= 3 && data[i] == 0 && data[i + 1] == 0 &&
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data[i + 2] == 3) {
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// Two rbsp bytes + the emulation byte.
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rbsp_buffer->AppendData(sps_bytes + i, 2);
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i += 3;
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} else {
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// Single rbsp byte.
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rbsp_buffer->AppendData(sps_bytes[i]);
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++i;
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}
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}
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return rbsp_buffer;
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}
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void WriteRbsp(const uint8_t* bytes, size_t length, rtc::Buffer* destination) {
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static const uint8_t kZerosInStartSequence = 2;
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static const uint8_t kEmulationByte = 0x03u;
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size_t num_consecutive_zeros = 0;
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destination->EnsureCapacity(destination->size() + length);
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for (size_t i = 0; i < length; ++i) {
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uint8_t byte = bytes[i];
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if (byte <= kEmulationByte &&
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num_consecutive_zeros >= kZerosInStartSequence) {
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// Need to escape.
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destination->AppendData(kEmulationByte);
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num_consecutive_zeros = 0;
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}
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destination->AppendData(byte);
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if (byte == 0) {
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++num_consecutive_zeros;
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} else {
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num_consecutive_zeros = 0;
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}
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}
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}
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} // namespace H264
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} // namespace webrtc
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