| /* |
| * Copyright 2017 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. |
| */ |
| |
| #ifndef ORTC_ORTCRTPSENDERADAPTER_H_ |
| #define ORTC_ORTCRTPSENDERADAPTER_H_ |
| |
| #include <memory> |
| |
| #include "api/ortc/ortcrtpsenderinterface.h" |
| #include "api/rtcerror.h" |
| #include "api/rtpparameters.h" |
| #include "ortc/rtptransportcontrolleradapter.h" |
| #include "pc/rtpsender.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| |
| namespace webrtc { |
| |
| // Implementation of OrtcRtpSenderInterface that works with RtpTransportAdapter, |
| // and wraps a VideoRtpSender/AudioRtpSender that's normally used with the |
| // PeerConnection. |
| // |
| // TODO(deadbeef): When BaseChannel is split apart into separate |
| // "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter |
| // object can be removed. |
| class OrtcRtpSenderAdapter : public OrtcRtpSenderInterface { |
| public: |
| // Wraps |wrapped_sender| in a proxy that will safely call methods on the |
| // correct thread. |
| static std::unique_ptr<OrtcRtpSenderInterface> CreateProxy( |
| std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender); |
| |
| // Should only be called by RtpTransportControllerAdapter. |
| OrtcRtpSenderAdapter(cricket::MediaType kind, |
| RtpTransportInterface* transport, |
| RtpTransportControllerAdapter* rtp_transport_controller); |
| ~OrtcRtpSenderAdapter() override; |
| |
| // OrtcRtpSenderInterface implementation. |
| RTCError SetTrack(MediaStreamTrackInterface* track) override; |
| rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override; |
| |
| RTCError SetTransport(RtpTransportInterface* transport) override; |
| RtpTransportInterface* GetTransport() const override; |
| |
| RTCError Send(const RtpParameters& parameters) override; |
| RtpParameters GetParameters() const override; |
| |
| cricket::MediaType GetKind() const override; |
| |
| // Used so that the RtpTransportControllerAdapter knows when it can |
| // deallocate resources allocated for this object. |
| sigslot::signal0<> SignalDestroyed; |
| |
| private: |
| void CreateInternalSender(); |
| |
| cricket::MediaType kind_; |
| RtpTransportInterface* transport_; |
| RtpTransportControllerAdapter* rtp_transport_controller_; |
| // Scoped refptr due to ref-counted interface, but we should be the only |
| // reference holder. |
| rtc::scoped_refptr<RtpSenderInternal> internal_sender_; |
| rtc::scoped_refptr<MediaStreamTrackInterface> track_; |
| RtpParameters last_applied_parameters_; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpSenderAdapter); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // ORTC_ORTCRTPSENDERADAPTER_H_ |