
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
81 lines
2.8 KiB
C++
81 lines
2.8 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/mediatypes.h"
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/proxy.h"
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#include "webrtc/api/rtpparameters.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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namespace webrtc {
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class RtpReceiverObserverInterface {
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public:
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// Note: Currently if there are multiple RtpReceivers of the same media type,
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// they will all call OnFirstPacketReceived at once.
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//
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// In the future, it's likely that an RtpReceiver will only call
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// OnFirstPacketReceived when a packet is received specifically for its
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// SSRC/mid.
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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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// Audio or video receiver?
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virtual cricket::MediaType media_type() 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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// Does not take ownership of observer.
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// Must call SetObserver(nullptr) before the observer is destroyed.
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virtual void SetObserver(RtpReceiverObserverInterface* observer) = 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_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
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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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END_PROXY_MAP()
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} // namespace webrtc
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#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_
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