| /* |
| * 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 <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 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<MediaStreamTrackInterface> 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_SIGNALING_PROXY_MAP(RtpSender) |
| PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, 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_SIGNALING_PROXY() |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_API_RTPSENDERINTERFACE_H_ |