e814a0dee0
This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
80 lines
2.8 KiB
C++
80 lines
2.8 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 WEBRTC_ORTC_ORTCRTPRECEIVERADAPTER_H_
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#define WEBRTC_ORTC_ORTCRTPRECEIVERADAPTER_H_
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#include <memory>
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#include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
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#include "webrtc/api/rtcerror.h"
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#include "webrtc/api/rtpparameters.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/thread.h"
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#include "webrtc/ortc/rtptransportcontrolleradapter.h"
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#include "webrtc/pc/rtpreceiver.h"
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namespace webrtc {
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// Implementation of OrtcRtpReceiverInterface that works with
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// RtpTransportAdapter, and wraps a VideoRtpReceiver/AudioRtpReceiver that's
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// normally used with the PeerConnection.
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//
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// TODO(deadbeef): When BaseChannel is split apart into separate
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// "RtpReceiver"/"RtpTransceiver"/"RtpReceiver"/"RtpReceiver" objects, this
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// adapter object can be removed.
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class OrtcRtpReceiverAdapter : public OrtcRtpReceiverInterface {
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public:
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// Wraps |wrapped_receiver| in a proxy that will safely call methods on the
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// correct thread.
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static std::unique_ptr<OrtcRtpReceiverInterface> CreateProxy(
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std::unique_ptr<OrtcRtpReceiverAdapter> wrapped_receiver);
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// Should only be called by RtpTransportControllerAdapter.
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OrtcRtpReceiverAdapter(
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cricket::MediaType kind,
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RtpTransportInterface* transport,
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RtpTransportControllerAdapter* rtp_transport_controller);
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~OrtcRtpReceiverAdapter() override;
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// OrtcRtpReceiverInterface implementation.
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rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override;
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RTCError SetTransport(RtpTransportInterface* transport) override;
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RtpTransportInterface* GetTransport() const override;
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RTCError Receive(const RtpParameters& parameters) override;
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RtpParameters GetParameters() const override;
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cricket::MediaType GetKind() const override;
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// Used so that the RtpTransportControllerAdapter knows when it can
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// deallocate resources allocated for this object.
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sigslot::signal0<> SignalDestroyed;
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private:
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void MaybeRecreateInternalReceiver();
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cricket::MediaType kind_;
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RtpTransportInterface* transport_;
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RtpTransportControllerAdapter* rtp_transport_controller_;
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// Scoped refptr due to ref-counted interface, but we should be the only
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// reference holder.
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rtc::scoped_refptr<RtpReceiverInternal> internal_receiver_;
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RtpParameters last_applied_parameters_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpReceiverAdapter);
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};
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} // namespace webrtc
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#endif // WEBRTC_ORTC_ORTCRTPRECEIVERADAPTER_H_
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