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5b4f075f9c
This is a reland off351c3408aReland history: The original CL broke tests in chromium which were manually tested in the first reland. Another small fix was added to the reland to fix a downstream bug, which caused separate tests to fail in chromium. These were not caught because the chromium trybot was down. These are temporarily disabled in chrome to allow this change to roll in. Original change's description: > Reland "Adds support for multiple or no media stream ids." > > This is a reland of1550292efe> > Original change's description: > > Adds support for multiple or no media stream ids. > > > > With Unified Plan SDP semantics, this adds support for specifying > > either no media stream ids or multiple media stream ids for a > > transceiver/sender/receiver. This includes serializing/deserializing > > SDPs with multiple a=msid lines in a m section, or an "a=msid:- > > <appdata>" line to indicate the no stream case. Note that this does > > not synchronize between multiple streams, this is still just supported > > based upon the first media stream id. > > > > Bug: webrtc:7932, webrtc:7933 > > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275 > > Reviewed-on: https://webrtc-review.googlesource.com/61341 > > Commit-Queue: Seth Hampson <shampson@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22611} > > Bug: webrtc:7932, webrtc:7933 > Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed > Reviewed-on: https://webrtc-review.googlesource.com/65560 > Commit-Queue: Seth Hampson <shampson@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22687} TBR=deadbeef@webrtc.org Bug: webrtc:7932, webrtc:7933 Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17 Reviewed-on: https://webrtc-review.googlesource.com/66280 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22699}
86 lines
3.1 KiB
C++
86 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/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() 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 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_CONSTMETHOD0(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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