/* * 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 RtpSenders // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface #ifndef WEBRTC_API_RTPSENDERINTERFACE_H_ #define WEBRTC_API_RTPSENDERINTERFACE_H_ #include #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 RtpSenderInterface : public rtc::RefCountInterface { public: // Returns true if successful in setting the track. // Fails if an audio track is set on a video RtpSender, or vice-versa. virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; virtual rtc::scoped_refptr track() const = 0; // Used to set the SSRC of the sender, once a local description has been set. // If |ssrc| is 0, this indiates that the sender should disconnect from the // underlying transport (this occurs if the sender isn't seen in a local // description). virtual void SetSsrc(uint32_t ssrc) = 0; virtual uint32_t ssrc() const = 0; // Audio or video sender? 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; // TODO(deadbeef): Support one sender having multiple stream ids. virtual void set_stream_id(const std::string& stream_id) = 0; virtual std::string stream_id() const = 0; virtual void Stop() = 0; virtual RtpParameters GetParameters() const = 0; virtual bool SetParameters(const RtpParameters& parameters) = 0; protected: virtual ~RtpSenderInterface() {} }; // Define proxy for RtpSenderInterface. BEGIN_PROXY_MAP(RtpSender) PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) PROXY_CONSTMETHOD0(rtc::scoped_refptr, track) PROXY_METHOD1(void, SetSsrc, uint32_t) PROXY_CONSTMETHOD0(uint32_t, ssrc) PROXY_CONSTMETHOD0(cricket::MediaType, media_type) PROXY_CONSTMETHOD0(std::string, id) PROXY_METHOD1(void, set_stream_id, const std::string&) PROXY_CONSTMETHOD0(std::string, stream_id) PROXY_METHOD0(void, Stop) PROXY_CONSTMETHOD0(RtpParameters, GetParameters); PROXY_METHOD1(bool, SetParameters, const RtpParameters&) END_PROXY() } // namespace webrtc #endif // WEBRTC_API_RTPSENDERINTERFACE_H_