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/*
* 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/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_