blob: 684afc4e676dbe1fb460eecf4c0e759e11bc0159 [file] [log] [blame]
* 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.
#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<> {
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&>
virtual bool IsWritable(bool rtcp) const = 0;
virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) = 0;
virtual bool SendRtcpPacket(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