webrtc_m130/webrtc/api/rtpreceiverinterface.h

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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// This file contains interfaces for RtpReceivers
// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
#ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_
#define WEBRTC_API_RTPRECEIVERINTERFACE_H_
#include <string>
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
namespace webrtc {
class RtpReceiverObserverInterface {
public:
// Note: Currently if there are multiple RtpReceivers of the same media type,
// they will all call OnFirstPacketReceived at once.
//
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
protected:
virtual ~RtpReceiverObserverInterface() {}
};
class RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
// Audio or video receiver?
virtual cricket::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
// The WebRTC specification only defines RTCRtpParameters in terms of senders,
// but this API also applies them to receivers, similar to ORTC:
// http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
virtual RtpParameters GetParameters() const = 0;
// Currently, doesn't support changing any parameters, but may in the future.
virtual bool SetParameters(const RtpParameters& parameters) = 0;
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
protected:
virtual ~RtpReceiverInterface() {}
};
// Define proxy for RtpReceiverInterface.
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
// are called on is an implementation detail.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
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
END_PROXY_MAP()
} // namespace webrtc
#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_