Files
platform-external-webrtc/webrtc/api/rtpreceiverinterface.h
Taylor Brandstetter ba29c6aac7 Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
Relanding again after fixing issue with RTC_DCHECKs.

This CL 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

Review URL: https://codereview.webrtc.org/2046173002 .

Cr-Commit-Position: refs/heads/master@{#13305}
2016-06-27 23:30:45 +00:00

80 lines
2.8 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/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/pc/mediasession.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;
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.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
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*);
END_SIGNALING_PROXY()
} // namespace webrtc
#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_