
This moves the implementation specific methods to separate classes (RtpSenderInternal/RtpReceiverInternal) so that the interface classes represent the interface that external applications can rely on. The reason this wasn't done earlier was that PeerConnection needed to store proxy pointers, but also needed to access implementation- specific methods on the underlying objects. This is now possible by using "RtpSenderProxyWithInternal<RtpSenderInternal>", which is a proxy that implements RtpSenderInterface but also provides direct access to an RtpSenderInternal. Review-Url: https://codereview.webrtc.org/2023373002 Cr-Commit-Position: refs/heads/master@{#13056}
73 lines
2.5 KiB
C++
73 lines
2.5 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 RtpSenders
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// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
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#ifndef WEBRTC_API_RTPSENDERINTERFACE_H_
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#define WEBRTC_API_RTPSENDERINTERFACE_H_
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#include <string>
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#include <vector>
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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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#include "webrtc/pc/mediasession.h"
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namespace webrtc {
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class RtpSenderInterface : public rtc::RefCountInterface {
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public:
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// Returns true if successful in setting the track.
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// Fails if an audio track is set on a video RtpSender, or vice-versa.
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virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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// Returns primary SSRC used by this sender for sending media.
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// Returns 0 if not yet determined.
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// TODO(deadbeef): Change to rtc::Optional.
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// TODO(deadbeef): Remove? With GetParameters this should be redundant.
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virtual uint32_t ssrc() const = 0;
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// Audio or video sender?
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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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virtual std::vector<std::string> stream_ids() const = 0;
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virtual RtpParameters GetParameters() const = 0;
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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protected:
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virtual ~RtpSenderInterface() {}
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};
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// Define proxy for RtpSenderInterface.
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BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_CONSTMETHOD0(uint32_t, ssrc)
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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(std::vector<std::string>, stream_ids)
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PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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END_SIGNALING_PROXY()
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} // namespace webrtc
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#endif // WEBRTC_API_RTPSENDERINTERFACE_H_
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