/*
 *  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.
 */
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_

#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"

namespace webrtc {
// Class to hold rtp packet with metadata for sender side.
class RtpPacketToSend : public rtp::Packet {
 public:
  explicit RtpPacketToSend(const ExtensionManager* extensions)
      : Packet(extensions) {}
  RtpPacketToSend(const RtpPacketToSend& packet) = default;
  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
      : Packet(extensions, capacity) {}

  RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
  // Time in local time base as close as it can to frame capture time.
  int64_t capture_time_ms() const { return capture_time_ms_; }
  void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }

 private:
  int64_t capture_time_ms_ = 0;
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_