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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
79 lines
2.8 KiB
C++
79 lines
2.8 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ORTC_ORTCRTPSENDERADAPTER_H_
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#define ORTC_ORTCRTPSENDERADAPTER_H_
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#include <memory>
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#include "api/ortc/ortcrtpsenderinterface.h"
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#include "api/rtcerror.h"
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#include "api/rtpparameters.h"
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#include "ortc/rtptransportcontrolleradapter.h"
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#include "pc/rtpsender.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/sigslot.h"
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namespace webrtc {
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// Implementation of OrtcRtpSenderInterface that works with RtpTransportAdapter,
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// and wraps a VideoRtpSender/AudioRtpSender that's normally used with the
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// PeerConnection.
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//
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// TODO(deadbeef): When BaseChannel is split apart into separate
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// "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter
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// object can be removed.
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class OrtcRtpSenderAdapter : public OrtcRtpSenderInterface {
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public:
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// Wraps |wrapped_sender| in a proxy that will safely call methods on the
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// correct thread.
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static std::unique_ptr<OrtcRtpSenderInterface> CreateProxy(
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std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender);
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// Should only be called by RtpTransportControllerAdapter.
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OrtcRtpSenderAdapter(cricket::MediaType kind,
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RtpTransportInterface* transport,
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RtpTransportControllerAdapter* rtp_transport_controller);
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~OrtcRtpSenderAdapter() override;
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// OrtcRtpSenderInterface implementation.
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RTCError SetTrack(MediaStreamTrackInterface* track) override;
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rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override;
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RTCError SetTransport(RtpTransportInterface* transport) override;
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RtpTransportInterface* GetTransport() const override;
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RTCError Send(const RtpParameters& parameters) override;
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RtpParameters GetParameters() const override;
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cricket::MediaType GetKind() const override;
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// Used so that the RtpTransportControllerAdapter knows when it can
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// deallocate resources allocated for this object.
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sigslot::signal0<> SignalDestroyed;
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private:
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void CreateInternalSender();
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cricket::MediaType kind_;
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RtpTransportInterface* transport_;
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RtpTransportControllerAdapter* rtp_transport_controller_;
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// Scoped refptr due to ref-counted interface, but we should be the only
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// reference holder.
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rtc::scoped_refptr<RtpSenderInternal> internal_sender_;
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rtc::scoped_refptr<MediaStreamTrackInterface> track_;
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RtpParameters last_applied_parameters_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpSenderAdapter);
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};
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} // namespace webrtc
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#endif // ORTC_ORTCRTPSENDERADAPTER_H_
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