e830e683c4
The TransportController will be replaced by the JsepTransportController and JsepTransport will be replace be JsepTransport2. The JsepTransportController will take the entire SessionDescription and handle the RtcpMux, Sdes and BUNDLE internally. The ownership model is also changed. The P2P layer transports are not ref-counted and will be owned by the JsepTransport2. In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport or SrtpTransport and it implements the public and internal interface by calling the transport underneath. Bug: webrtc:8587 Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df Reviewed-on: https://webrtc-review.googlesource.com/59586 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22693}
95 lines
3.5 KiB
C++
95 lines
3.5 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTPTRANSPORTINTERNAL_H_
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#define PC_RTPTRANSPORTINTERNAL_H_
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#include <string>
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#include "api/ortc/srtptransportinterface.h"
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#include "api/umametrics.h"
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#include "p2p/base/icetransportinternal.h"
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#include "rtc_base/networkroute.h"
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#include "rtc_base/sigslot.h"
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#include "rtc_base/sslstreamadapter.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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struct PacketTime;
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} // namespace rtc
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namespace webrtc {
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// This represents the internal interface beneath SrtpTransportInterface;
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// it is not accessible to API consumers but is accessible to internal classes
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// in order to send and receive RTP and RTCP packets belonging to a single RTP
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// session. Additional convenience and configuration methods are also provided.
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class RtpTransportInternal : public SrtpTransportInterface,
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public sigslot::has_slots<> {
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public:
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virtual void SetRtcpMuxEnabled(bool enable) = 0;
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// TODO(zstein): Remove PacketTransport setters. Clients should pass these
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// in to constructors instead and construct a new RtpTransportInternal instead
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// of updating them.
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virtual bool rtcp_mux_enabled() const = 0;
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virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0;
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virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0;
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virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0;
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virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0;
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virtual bool IsReadyToSend() const = 0;
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// Called whenever a transport's ready-to-send state changes. The argument
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// is true if all used transports are ready to send. This is more specific
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// than just "writable"; it means the last send didn't return ENOTCONN.
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sigslot::signal1<bool> SignalReadyToSend;
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// TODO(zstein): Consider having two signals - RtpPacketReceived and
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// RtcpPacketReceived.
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// The first argument is true for RTCP packets and false for RTP packets.
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sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
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SignalPacketReceived;
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// Called whenever the network route of the P2P layer transport changes.
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// The argument is an optional network route.
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sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
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// Called whenever a transport's writable state might change. The argument is
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// true if the transport is writable, otherwise it is false.
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sigslot::signal1<bool> SignalWritableState;
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sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
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virtual bool IsWritable(bool rtcp) const = 0;
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// TODO(zhihuang): Pass the |packet| by copy so that the original data
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// wouldn't be modified.
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virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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virtual bool IsSrtpActive() const = 0;
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virtual void SetMetricsObserver(
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rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) = 0;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTINTERNAL_H_
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