/*
 *  Copyright 2017 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_PC_RTPTRANSPORTINTERNAL_H_
#define WEBRTC_PC_RTPTRANSPORTINTERNAL_H_

#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/rtc_base/sigslot.h"

namespace rtc {
class CopyOnWriteBuffer;
struct PacketOptions;
struct PacketTime;
}  // namespace rtc

namespace webrtc {

// This represents the internal interface beneath RtpTransportInterface;
// it is not accessible to API consumers but is accessible to internal classes
// in order to send and receive RTP and RTCP packets belonging to a single RTP
// session. Additional convenience and configuration methods are also provided.
class RtpTransportInternal : public RtpTransportInterface,
                             public sigslot::has_slots<> {
 public:
  virtual void SetRtcpMuxEnabled(bool enable) = 0;

  // TODO(zstein): Remove PacketTransport setters. Clients should pass these
  // in to constructors instead and construct a new RtpTransportInternal instead
  // of updating them.

  virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0;
  virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0;

  virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0;
  virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0;

  // Called whenever a transport's ready-to-send state changes. The argument
  // is true if all used transports are ready to send. This is more specific
  // than just "writable"; it means the last send didn't return ENOTCONN.
  sigslot::signal1<bool> SignalReadyToSend;

  // TODO(zstein): Consider having two signals - RtpPacketReceived and
  // RtcpPacketReceived.
  // The first argument is true for RTCP packets and false for RTP packets.
  sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
      SignalPacketReceived;

  virtual bool IsWritable(bool rtcp) const = 0;

  virtual bool SendPacket(bool rtcp,
                          rtc::CopyOnWriteBuffer* packet,
                          const rtc::PacketOptions& options,
                          int flags) = 0;

  virtual bool HandlesPayloadType(int payload_type) const = 0;

  virtual void AddHandledPayloadType(int payload_type) = 0;
};

}  // namespace webrtc

#endif  // WEBRTC_PC_RTPTRANSPORTINTERNAL_H_