chromium/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_generic.cc

/*
 *  Copyright (c) 2019 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 "modules/rtp_rtcp/source/video_rtp_depacketizer_generic.h"

#include <stddef.h>
#include <stdint.h>

#include <utility>

#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace {
constexpr uint8_t kKeyFrameBit = …;
constexpr uint8_t kFirstPacketBit = …;
// If this bit is set, there will be an extended header contained in this
// packet. This was added later so old clients will not send this.
constexpr uint8_t kExtendedHeaderBit = …;

constexpr size_t kGenericHeaderLength = …;
constexpr size_t kExtendedHeaderLength = …;
}  // namespace

absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerGeneric::Parse(rtc::CopyOnWriteBuffer rtp_payload) { … }
}  // namespace webrtc