2015-09-24 16:47:53 -07:00
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/*
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2016-02-10 07:54:43 -08:00
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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2015-09-24 16:47:53 -07:00
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*
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2016-02-10 07:54:43 -08:00
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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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2015-09-24 16:47:53 -07:00
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*/
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2015-09-28 16:53:55 -07:00
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// This file contains interfaces for RtpReceivers
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// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
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2016-02-10 10:53:12 +01:00
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#ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
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#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
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2015-09-28 16:53:55 -07:00
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#include <string>
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2017-01-23 04:56:25 -08:00
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#include "webrtc/api/mediatypes.h"
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2016-02-10 10:53:12 +01:00
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/proxy.h"
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2016-06-27 16:30:35 -07:00
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#include "webrtc/api/rtpparameters.h"
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2015-09-28 16:53:55 -07:00
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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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2016-06-14 11:47:14 -07:00
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class RtpReceiverObserverInterface {
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public:
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// Note: Currently if there are multiple RtpReceivers of the same media type,
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// they will all call OnFirstPacketReceived at once.
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//
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// In the future, it's likely that an RtpReceiver will only call
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// OnFirstPacketReceived when a packet is received specifically for its
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// SSRC/mid.
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virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
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protected:
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virtual ~RtpReceiverObserverInterface() {}
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};
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2015-09-28 16:53:55 -07:00
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class RtpReceiverInterface : public rtc::RefCountInterface {
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public:
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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2016-06-27 16:30:35 -07:00
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// Audio or video receiver?
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virtual cricket::MediaType media_type() const = 0;
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2015-09-28 16:53:55 -07:00
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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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2016-05-16 11:40:30 -07:00
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// The WebRTC specification only defines RTCRtpParameters in terms of senders,
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// but this API also applies them to receivers, similar to ORTC:
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// http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
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virtual RtpParameters GetParameters() const = 0;
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// Currently, doesn't support changing any parameters, but may in the future.
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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2016-06-27 16:30:35 -07:00
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// Does not take ownership of observer.
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// Must call SetObserver(nullptr) before the observer is destroyed.
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virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
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2015-09-28 16:53:55 -07:00
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protected:
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virtual ~RtpReceiverInterface() {}
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};
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// Define proxy for RtpReceiverInterface.
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2017-02-08 01:38:21 -08:00
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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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2016-04-15 03:49:07 -07:00
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BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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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(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
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Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
2017-04-10 04:38:13 -07:00
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END_PROXY_MAP()
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2015-09-28 16:53:55 -07:00
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} // namespace webrtc
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2016-02-10 10:53:12 +01:00
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#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_
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