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Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
87 lines
3.1 KiB
C++
87 lines
3.1 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 API_RTPSENDERINTERFACE_H_
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#define API_RTPSENDERINTERFACE_H_
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#include <string>
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#include <vector>
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#include "api/dtmfsenderinterface.h"
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#include "api/mediastreaminterface.h"
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#include "api/mediatypes.h"
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#include "api/proxy.h"
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#include "api/rtcerror.h"
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#include "api/rtpparameters.h"
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#include "rtc_base/deprecation.h"
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#include "rtc_base/refcount.h"
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#include "rtc_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 media stream ids associated with this sender's track.
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// These are signalled in the SDP so that the remote side can associate
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// tracks.
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virtual std::vector<std::string> stream_ids() const = 0;
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virtual RtpParameters GetParameters() = 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 RTCError 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_METHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(RTCError, 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 // API_RTPSENDERINTERFACE_H_
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