Files
platform-external-webrtc/webrtc/api/rtpreceiverinterface.h
deadbeef 1a7162dbc9 Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.

Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}

TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
2016-06-24 21:13:14 +00:00

70 lines
2.3 KiB
C++

/*
* 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/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/pc/mediasession.h"
namespace webrtc {
class RtpReceiverObserverInterface {
public:
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;
// 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;
virtual bool SetParameters(const RtpParameters& parameters) = 0;
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
virtual cricket::MediaType media_type() = 0;
protected:
virtual ~RtpReceiverInterface() {}
};
// Define proxy for RtpReceiverInterface.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
PROXY_METHOD0(cricket::MediaType, media_type);
END_SIGNALING_PROXY()
} // namespace webrtc
#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_