| /* |
| * 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 CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |
| #define CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |
| |
| #include <memory> |
| |
| #include "call/rtp_demuxer.h" |
| #include "call/rtp_stream_receiver_controller_interface.h" |
| #include "rtc_base/critical_section.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. |
| bool 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 |
| // 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. |
| rtc::CriticalSection lock_; |
| RtpDemuxer demuxer_ RTC_GUARDED_BY(&lock_); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |