chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

/*
 *  Copyright (c) 2014 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.
 */

#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_

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

#include <deque>
#include <memory>
#include <queue>

#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "rtc_base/buffer.h"

namespace webrtc {

// Bit masks for NAL (F, NRI, Type) indicators.
constexpr uint8_t kH264FBit = …;
constexpr uint8_t kH264NriMask = …;
constexpr uint8_t kH264TypeMask = …;

// Bit masks for FU (A and B) headers.
constexpr uint8_t kH264SBit = …;
constexpr uint8_t kH264EBit = …;
constexpr uint8_t kH264RBit = …;

class RtpPacketizerH264 : public RtpPacketizer { … };
}  // namespace webrtc
#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_