
This reverts commit e357a4dd4e3b015f8281813f246de793589bd537. Reason for revert: Looks like it's breaking some downstream projects. Original change's description: > Move stats ID generation from SSRC to local ID > > This generates stats IDs for Track stats (which > represents stats on the attachment of a track to > a PeerConnection) from being SSRC-based to being > based on an ID that is allocated when connecting the > track to the PC. > > This is a prerequisite to generating stats before > the PeerConnection is connected. > > Bug: webrtc:8673 > Change-Id: I82f6e521646b0c92b3af4dffb2cdee75e6dc10d4 > Reviewed-on: https://webrtc-review.googlesource.com/38360 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21582} TBR=solenberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org Change-Id: I621c10236c02be01d82f4660168f0323b85e24af No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8673 Reviewed-on: https://webrtc-review.googlesource.com/38681 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21586}
85 lines
3.1 KiB
C++
85 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/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 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 // API_RTPSENDERINTERFACE_H_
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