365381fdf1
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
122 lines
4.6 KiB
C++
122 lines
4.6 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 ORTC_RTPTRANSPORTADAPTER_H_
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#define ORTC_RTPTRANSPORTADAPTER_H_
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#include <memory>
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#include <vector>
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#include "api/rtcerror.h"
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#include "media/base/streamparams.h"
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#include "ortc/rtptransportcontrolleradapter.h"
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#include "pc/channel.h"
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#include "pc/rtptransportinternaladapter.h"
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#include "pc/srtptransport.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/sigslot.h"
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namespace webrtc {
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// This class is a wrapper over an RtpTransport or an SrtpTransport. The base
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// class RtpTransportInternalAdapter keeps a raw pointer, |transport_|, of the
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// transport object and implements both the public SrtpTransportInterface and
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// RtpTransport internal interface by calling the |transport_| underneath.
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//
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// This adapter can be used as an unencrypted RTP transport or an SrtpTransport
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// with RtpSenderAdapter, RtpReceiverAdapter, and RtpTransportControllerAdapter.
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//
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// TODO(deadbeef): When BaseChannel is split apart into separate
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// "RtpTransport"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this
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// adapter object can be removed.
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class RtpTransportAdapter : public RtpTransportInternalAdapter {
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public:
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// |rtp| can't be null. |rtcp| can if RTCP muxing is used immediately (meaning
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// |rtcp_parameters.mux| is also true).
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static RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxied(
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const RtpTransportParameters& rtcp_parameters,
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PacketTransportInterface* rtp,
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PacketTransportInterface* rtcp,
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RtpTransportControllerAdapter* rtp_transport_controller);
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static RTCErrorOr<std::unique_ptr<SrtpTransportInterface>> CreateSrtpProxied(
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const RtpTransportParameters& rtcp_parameters,
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PacketTransportInterface* rtp,
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PacketTransportInterface* rtcp,
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RtpTransportControllerAdapter* rtp_transport_controller);
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~RtpTransportAdapter() override;
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// RtpTransportInterface implementation.
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PacketTransportInterface* GetRtpPacketTransport() const override {
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return rtp_packet_transport_;
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}
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PacketTransportInterface* GetRtcpPacketTransport() const override {
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return rtcp_packet_transport_;
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}
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RTCError SetParameters(const RtpTransportParameters& parameters) override;
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RtpTransportParameters GetParameters() const override { return parameters_; }
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// SRTP specific implementation.
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RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override;
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RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override;
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// Methods used internally by OrtcFactory.
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RtpTransportControllerAdapter* rtp_transport_controller() {
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return rtp_transport_controller_;
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}
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void TakeOwnershipOfRtpTransportController(
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std::unique_ptr<RtpTransportControllerInterface> controller);
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// Used by RtpTransportControllerAdapter to tell when it should stop
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// returning this transport from GetTransports().
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sigslot::signal1<RtpTransportAdapter*> SignalDestroyed;
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bool IsSrtpActive() const override { return transport_->IsSrtpActive(); }
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protected:
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RtpTransportAdapter* GetInternal() override { return this; }
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private:
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RtpTransportAdapter(const RtcpParameters& rtcp_params,
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PacketTransportInterface* rtp,
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PacketTransportInterface* rtcp,
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RtpTransportControllerAdapter* rtp_transport_controller,
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bool is_srtp_transport);
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void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& time) {
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SignalRtcpPacketReceived(packet, time);
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}
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void OnWritableState(bool writable) { SignalWritableState(writable); }
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PacketTransportInterface* rtp_packet_transport_ = nullptr;
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PacketTransportInterface* rtcp_packet_transport_ = nullptr;
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RtpTransportControllerAdapter* const rtp_transport_controller_ = nullptr;
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// Non-null if this class owns the transport controller.
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std::unique_ptr<RtpTransportControllerInterface>
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owned_rtp_transport_controller_;
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RtpTransportParameters parameters_;
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// Only one of them is non-null;
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std::unique_ptr<RtpTransport> unencrypted_rtp_transport_;
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std::unique_ptr<SrtpTransport> srtp_transport_;
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rtc::Thread* network_thread_ = nullptr;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportAdapter);
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};
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} // namespace webrtc
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#endif // ORTC_RTPTRANSPORTADAPTER_H_
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