| Index: webrtc/ortc/ortcrtpsenderadapter.h
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| diff --git a/webrtc/ortc/ortcrtpsenderadapter.h b/webrtc/ortc/ortcrtpsenderadapter.h
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| new file mode 100644
|
| index 0000000000000000000000000000000000000000..9b60f1530097c72899af00f5e8bf357ef9aaa964
|
| --- /dev/null
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| +++ b/webrtc/ortc/ortcrtpsenderadapter.h
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| @@ -0,0 +1,79 @@
|
| +/*
|
| + * 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 WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_
|
| +#define WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_
|
| +
|
| +#include <memory>
|
| +
|
| +#include "webrtc/api/ortc/ortcrtpsenderinterface.h"
|
| +#include "webrtc/api/rtcerror.h"
|
| +#include "webrtc/api/rtpparameters.h"
|
| +#include "webrtc/base/constructormagic.h"
|
| +#include "webrtc/base/sigslot.h"
|
| +#include "webrtc/ortc/rtptransportcontrolleradapter.h"
|
| +#include "webrtc/pc/rtpsender.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +// Implementation of OrtcRtpSenderInterface that works with RtpTransportAdapter,
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| +// and wraps a VideoRtpSender/AudioRtpSender that's normally used with the
|
| +// 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 {
|
| + public:
|
| + // Wraps |wrapped_sender| in a proxy that will safely call methods on the
|
| + // correct thread.
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| + static std::unique_ptr<OrtcRtpSenderInterface> CreateProxy(
|
| + std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender);
|
| +
|
| + // 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);
|
| + ~OrtcRtpSenderAdapter() override;
|
| +
|
| + // OrtcRtpSenderInterface implementation.
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| + RTCError SetTrack(MediaStreamTrackInterface* track) override;
|
| + rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override;
|
| +
|
| + RTCError SetTransport(RtpTransportInterface* transport) override;
|
| + RtpTransportInterface* GetTransport() const override;
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| +
|
| + RTCError Send(const RtpParameters& parameters) override;
|
| + 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;
|
| +
|
| + 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_;
|
| + // Scoped refptr due to ref-counted interface, but we should be the only
|
| + // 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);
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_ORTC_ORTCRTPSENDERADAPTER_H_
|
|
|