/* * 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 API_RTPSENDERINTERFACE_H_ #define API_RTPSENDERINTERFACE_H_ #include #include #include "api/dtmfsenderinterface.h" #include "api/mediastreaminterface.h" #include "api/mediatypes.h" #include "api/proxy.h" #include "api/rtcerror.h" #include "api/rtpparameters.h" #include "rtc_base/deprecation.h" #include "rtc_base/refcount.h" #include "rtc_base/scoped_ref_ptr.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; // Returns primary SSRC used by this sender for sending media. // Returns 0 if not yet determined. // TODO(deadbeef): Change to absl::optional. // TODO(deadbeef): Remove? With GetParameters this should be redundant. 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; // Returns a list of media stream ids associated with this sender's track. // These are signalled in the SDP so that the remote side can associate // tracks. virtual std::vector stream_ids() const = 0; virtual RtpParameters GetParameters() = 0; // Note that only a subset of the parameters can currently be changed. See // rtpparameters.h // The encodings are in increasing quality order for simulcast. virtual RTCError SetParameters(const RtpParameters& parameters) = 0; // Returns null for a video sender. virtual rtc::scoped_refptr GetDtmfSender() const = 0; protected: ~RtpSenderInterface() override = default; }; // Define proxy for RtpSenderInterface. // 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(RtpSender) PROXY_SIGNALING_THREAD_DESTRUCTOR() PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) PROXY_CONSTMETHOD0(rtc::scoped_refptr, track) PROXY_CONSTMETHOD0(uint32_t, ssrc) PROXY_CONSTMETHOD0(cricket::MediaType, media_type) PROXY_CONSTMETHOD0(std::string, id) PROXY_CONSTMETHOD0(std::vector, stream_ids) PROXY_METHOD0(RtpParameters, GetParameters); PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&) PROXY_CONSTMETHOD0(rtc::scoped_refptr, GetDtmfSender); END_PROXY_MAP() } // namespace webrtc #endif // API_RTPSENDERINTERFACE_H_