blob: bf052b3a23800b9d929137633e481c4bed95fd2d [file] [log] [blame]
* 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/base/buffer.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_video/h264/sps_parser.h"
namespace rtc {
class BitBuffer;
namespace webrtc {
// A class that can parse an SPS block of a NAL unit and if necessary
// creates a copy with updated settings to allow for faster decoding for streams
// that use picture order count type 0. Streams in that format incur additional
// delay because it allows decode order to differ from render order.
// The mechanism used is to rewrite (edit or add) the SPS's VUI to contain
// restrictions on the maximum number of reordered pictures. This reduces
// latency significantly, though it still adds about a frame of latency to
// decoding.
class SpsVuiRewriter : private SpsParser {
enum class ParseResult { kFailure, kPocOk, kVuiOk, kVuiRewritten };
// Parses an SPS block and if necessary copies it and rewrites the VUI.
// Returns kFailure on failure, kParseOk if parsing succeeded and no update
// was necessary and kParsedAndModified if an updated copy of buffer was
// written to destination. destination may be populated with some data even if
// no rewrite was necessary, but the end offset should remain unchanged.
// Unless parsing fails, the sps parameter will be populated with the parsed
// SPS state. This function assumes that any previous headers
// (NALU start, type, Stap-A, etc) have already been parsed and that RBSP
// decoding has been performed.
static ParseResult ParseAndRewriteSps(const uint8_t* buffer,
size_t length,
rtc::Optional<SpsParser::SpsState>* sps,
rtc::Buffer* destination);
} // namespace webrtc