mirror of
https://github.com/mollyim/webrtc.git
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ee2388f3f0
This reverts commit 56bae8ded3.
Reason for revert: Speculative revert. This CL is suspect of making Chrome trybots fail the following test, preventing rolls:
external/wpt/webrtc/RTCPeerConnection-track-stats.https.html
Some failed roll attempts:
https://chromium-review.googlesource.com/c/chromium/src/+/921421
https://chromium-review.googlesource.com/c/chromium/src/+/921422
https://chromium-review.googlesource.com/c/chromium/src/+/921781
Some failed bot runs:
https://ci.chromium.org/buildbot/tryserver.chromium.linux/linux_chromium_rel_ng/647669
https://ci.chromium.org/buildbot/tryserver.chromium.win/win7_chromium_rel_ng/103786
Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
>
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
Change-Id: I21ce2109087d7b2d9470471ee9a6757f904296d2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8764
Reviewed-on: https://webrtc-review.googlesource.com/54000
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22036}
93 lines
3.5 KiB
C++
93 lines
3.5 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 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 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 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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// Returns an ID that changes every time SetTrack() is called, but
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// otherwise remains constant. Used to generate IDs for stats.
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// The special value zero means that no track is attached.
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// TODO(hta): Remove default implementation when callers have updated,
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// or move function to an internal interface.
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virtual int AttachmentId() const { return 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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PROXY_CONSTMETHOD0(int, AttachmentId);
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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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