| Index: webrtc/call/rtp_transport_controller_receive.h
|
| diff --git a/webrtc/call/rtp_transport_controller_receive.h b/webrtc/call/rtp_transport_controller_receive.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..24bc894e02f797e9bad0479305d86578cb0df5bf
|
| --- /dev/null
|
| +++ b/webrtc/call/rtp_transport_controller_receive.h
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| @@ -0,0 +1,67 @@
|
| +/*
|
| + * Copyright (c) 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_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_
|
| +#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_
|
| +
|
| +#include <memory>
|
| +
|
| +#include "webrtc/base/array_view.h"
|
| +#include "webrtc/base/criticalsection.h"
|
| +
|
| +#include "webrtc/call/rtp_demuxer.h"
|
| +#include "webrtc/call/rtp_transport_controller_receive_interface.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +class ReceiveSideCongestionController;
|
| +class RtpPacketReceived;
|
| +
|
| +// This class represents the RTP receive parsing and demuxing, for a
|
| +// single RTP session. It isn't thread aware, leaving responsibility
|
| +// of multithreading issues to the user of this class.
|
| +// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP
|
| +// and not leave any RTCP processing to individual receive streams.
|
| +// TODO(nisse): Extract packet processing, including parsing and
|
| +// demuxing, into a separate RtpSessionReceiver classes.
|
| +class RtpTransportControllerReceive
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| + : public RtpTransportControllerReceiveInterface {
|
| + public:
|
| + // Implements RtpTransportControllerReceiveInterface.
|
| + std::unique_ptr<RtpTransportReceiver> CreateReceiver(
|
| + uint32_t ssrc,
|
| + RtpPacketSinkInterface* sink) override;
|
| +
|
| + // Thread-safe wrappers for the corresponding RtpDemuxer methods.
|
| + void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override;
|
| + size_t RemoveSink(const RtpPacketSinkInterface* sink) override;
|
| +
|
| + // TODO(nisse): Not yet responsible for parsing.
|
| + bool OnRtpPacket(const RtpPacketReceived& packet);
|
| +
|
| + private:
|
| + class Receiver : public RtpTransportReceiver {
|
| + public:
|
| + Receiver(RtpTransportControllerReceive* transport,
|
| + uint32_t ssrc, RtpPacketSinkInterface* sink);
|
| +
|
| + ~Receiver() override;
|
| +
|
| + private:
|
| + RtpTransportControllerReceive *transport_;
|
| + RtpPacketSinkInterface* sink_;
|
| + };
|
| +
|
| + rtc::CriticalSection lock_;
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| + RtpDemuxer demuxer_ GUARDED_BY(&lock_);
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_
|
|
|