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

/*
 *  Copyright (c) 2013 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_VIDEO_GENERIC_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_

#include <stdint.h>

#include <vector>

#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_format.h"

namespace webrtc {

class RtpPacketToSend;
struct RTPVideoHeader;

namespace RtpFormatVideoGeneric {
static const uint8_t kKeyFrameBit = …;
static const 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.
static const uint8_t kExtendedHeaderBit = …;
}  // namespace RtpFormatVideoGeneric

class RtpPacketizerGeneric : public RtpPacketizer { … };
}  // namespace webrtc
#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_