
Reason for revert: Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland. Original issue's description: > Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. > > This eliminates the need for the extra layer of indirection provided by > mediastreamprovider.h. It will thus make it easier to implement new > functionality in RtpSender/RtpReceiver. > > It also brings us one step closer to the end goal of combining "senders" > and "send streams". Currently the sender still needs to go through the > BaseChannel and MediaChannel, using an SSRC as a key. > > R=pthatcher@webrtc.org > > Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c > Cr-Commit-Position: refs/heads/master@{#13285} TBR=pthatcher@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2099843003 Cr-Commit-Position: refs/heads/master@{#13286}
70 lines
2.3 KiB
C++
70 lines
2.3 KiB
C++
/*
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* Copyright 2015 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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// This file contains interfaces for RtpReceivers
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// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
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#ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
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#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
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#include <string>
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/proxy.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/pc/mediasession.h"
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namespace webrtc {
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class RtpReceiverObserverInterface {
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public:
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virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
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protected:
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virtual ~RtpReceiverObserverInterface() {}
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};
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class RtpReceiverInterface : public rtc::RefCountInterface {
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public:
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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// Not to be confused with "mid", this is a field we can temporarily use
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// to uniquely identify a receiver until we implement Unified Plan SDP.
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virtual std::string id() const = 0;
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// The WebRTC specification only defines RTCRtpParameters in terms of senders,
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// but this API also applies them to receivers, similar to ORTC:
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// http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
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virtual RtpParameters GetParameters() const = 0;
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
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virtual cricket::MediaType media_type() = 0;
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protected:
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virtual ~RtpReceiverInterface() {}
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};
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// Define proxy for RtpReceiverInterface.
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BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_CONSTMETHOD0(std::string, id)
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PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
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PROXY_METHOD0(cricket::MediaType, media_type);
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END_SIGNALING_PROXY()
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} // namespace webrtc
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#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_
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