| Index: webrtc/call/rtp_stream_receiver_controller.h
|
| diff --git a/webrtc/call/rtp_stream_receiver_controller.h b/webrtc/call/rtp_stream_receiver_controller.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..5c8ed6775e8739b59e3db79c967f3df7e7ab10cb
|
| --- /dev/null
|
| +++ b/webrtc/call/rtp_stream_receiver_controller.h
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| @@ -0,0 +1,72 @@
|
| +/*
|
| + * 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_STREAM_RECEIVER_CONTROLLER_H_
|
| +#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
|
| +
|
| +#include <memory>
|
| +
|
| +#include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/call/rtp_demuxer.h"
|
| +#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +class RtpPacketReceived;
|
| +
|
| +// This class represents the RTP receive parsing and demuxing, for a
|
| +// single RTP session.
|
| +// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP
|
| +// and not leave any RTCP processing to individual receive streams.
|
| +// TODO(nisse): Extract per-packet processing, including parsing and
|
| +// demuxing, into a separate class.
|
| +class RtpStreamReceiverController
|
| + : public RtpStreamReceiverControllerInterface {
|
| + public:
|
| + RtpStreamReceiverController();
|
| + ~RtpStreamReceiverController() override;
|
| +
|
| + // Implements RtpStreamReceiverControllerInterface.
|
| + std::unique_ptr<RtpStreamReceiverInterface> 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 RtpStreamReceiverInterface {
|
| + public:
|
| + Receiver(RtpStreamReceiverController* controller,
|
| + uint32_t ssrc,
|
| + RtpPacketSinkInterface* sink);
|
| +
|
| + ~Receiver() override;
|
| +
|
| + private:
|
| + RtpStreamReceiverController* const controller_;
|
| + RtpPacketSinkInterface* const sink_;
|
| + };
|
| +
|
| + // TODO(nisse): Move to a TaskQueue for synchronization. When used
|
| + // by Call, we expect construction and all methods but OnRtpPacket
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| + // to be called on the same thread, and OnRtpPacket to be called
|
| + // by a single, but possibly distinct, thread. But applications not
|
| + // using Call may have use threads differently.
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| + rtc::CriticalSection lock_;
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| + RtpDemuxer demuxer_ GUARDED_BY(&lock_);
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
|
|
|