b10f32f9b2
Many of these interfaces are not intuitive, or are the way they are for complex historical reasons, so it would be nice to document these things for future developers. Also, many nonstandard things (such as RTCConfiguration options) were not documented at all before this CL. BUG=webrtc:7131 TBR=pthatcher@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2680273002 Cr-Commit-Position: refs/heads/master@{#16485}
85 lines
3.2 KiB
C++
85 lines
3.2 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/dtmfsenderinterface.h"
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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 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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// Returns a list of streams associated with this sender's track. Although we
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// only support one track per stream, in theory the API allows for multiple.
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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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// Note that only a subset of the parameters can currently be changed. See
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// rtpparameters.h
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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// Returns null for a video sender.
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virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 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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// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
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// are called on is an implementation detail.
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BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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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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PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
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END_PROXY_MAP()
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} // namespace webrtc
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#endif // WEBRTC_API_RTPSENDERINTERFACE_H_
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