mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
e41c433502
Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
110 lines
4.3 KiB
C++
110 lines
4.3 KiB
C++
/*
|
|
* 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 PC_RTPTRANSPORTINTERNAL_H_
|
|
#define PC_RTPTRANSPORTINTERNAL_H_
|
|
|
|
#include <string>
|
|
|
|
#include "api/ortc/srtptransportinterface.h"
|
|
#include "call/rtp_demuxer.h"
|
|
#include "p2p/base/icetransportinternal.h"
|
|
#include "pc/sessiondescription.h"
|
|
#include "rtc_base/networkroute.h"
|
|
#include "rtc_base/sslstreamadapter.h"
|
|
#include "rtc_base/third_party/sigslot/sigslot.h"
|
|
|
|
namespace rtc {
|
|
class CopyOnWriteBuffer;
|
|
struct PacketOptions;
|
|
struct PacketTime;
|
|
} // namespace rtc
|
|
|
|
namespace webrtc {
|
|
|
|
// This represents the internal interface beneath SrtpTransportInterface;
|
|
// 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 SrtpTransportInterface,
|
|
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 bool rtcp_mux_enabled() const = 0;
|
|
|
|
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;
|
|
|
|
virtual bool IsReadyToSend() const = 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;
|
|
|
|
// Called whenever an RTCP packet is received. There is no equivalent signal
|
|
// for RTP packets because they would be forwarded to the BaseChannel through
|
|
// the RtpDemuxer callback.
|
|
sigslot::signal2<rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
|
|
SignalRtcpPacketReceived;
|
|
|
|
// Called whenever the network route of the P2P layer transport changes.
|
|
// The argument is an optional network route.
|
|
sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
|
|
|
|
// Called whenever a transport's writable state might change. The argument is
|
|
// true if the transport is writable, otherwise it is false.
|
|
sigslot::signal1<bool> SignalWritableState;
|
|
|
|
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
|
|
|
|
virtual bool IsWritable(bool rtcp) const = 0;
|
|
|
|
// TODO(zhihuang): Pass the |packet| by copy so that the original data
|
|
// wouldn't be modified.
|
|
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;
|
|
|
|
// This method updates the RTP header extension map so that the RTP transport
|
|
// can parse the received packets and identify the MID. This is called by the
|
|
// BaseChannel when setting the content description.
|
|
//
|
|
// TODO(zhihuang): Merging and replacing following methods handling header
|
|
// extensions with SetParameters:
|
|
// UpdateRtpHeaderExtensionMap,
|
|
// UpdateSendEncryptedHeaderExtensionIds,
|
|
// UpdateRecvEncryptedHeaderExtensionIds,
|
|
// CacheRtpAbsSendTimeHeaderExtension,
|
|
virtual void UpdateRtpHeaderExtensionMap(
|
|
const cricket::RtpHeaderExtensions& header_extensions) = 0;
|
|
|
|
virtual bool IsSrtpActive() const = 0;
|
|
|
|
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
|
|
RtpPacketSinkInterface* sink) = 0;
|
|
|
|
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // PC_RTPTRANSPORTINTERNAL_H_
|