| /* | 
 |  *  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. | 
 |  */ | 
 |  | 
 | #include "call/rtp_stream_receiver_controller.h" | 
 |  | 
 | #include <cstdint> | 
 | #include <memory> | 
 |  | 
 | #include "api/sequence_checker.h" | 
 | #include "call/rtp_packet_sink_interface.h" | 
 | #include "call/rtp_stream_receiver_controller_interface.h" | 
 | #include "rtc_base/logging.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | RtpStreamReceiverController::Receiver::Receiver( | 
 |     RtpStreamReceiverController* controller, | 
 |     uint32_t ssrc, | 
 |     RtpPacketSinkInterface* sink) | 
 |     : controller_(controller), sink_(sink) { | 
 |   const bool sink_added = controller_->AddSink(ssrc, sink_); | 
 |   if (!sink_added) { | 
 |     RTC_LOG(LS_ERROR) | 
 |         << "RtpStreamReceiverController::Receiver::Receiver: Sink " | 
 |            "could not be added for SSRC=" | 
 |         << ssrc << "."; | 
 |   } | 
 | } | 
 |  | 
 | RtpStreamReceiverController::Receiver::~Receiver() { | 
 |   // This may fail, if corresponding AddSink in the constructor failed. | 
 |   controller_->RemoveSink(sink_); | 
 | } | 
 |  | 
 | RtpStreamReceiverController::RtpStreamReceiverController() {} | 
 |  | 
 | RtpStreamReceiverController::~RtpStreamReceiverController() = default; | 
 |  | 
 | std::unique_ptr<RtpStreamReceiverInterface> | 
 | RtpStreamReceiverController::CreateReceiver(uint32_t ssrc, | 
 |                                             RtpPacketSinkInterface* sink) { | 
 |   return std::make_unique<Receiver>(this, ssrc, sink); | 
 | } | 
 |  | 
 | bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) { | 
 |   RTC_DCHECK_RUN_ON(&demuxer_sequence_); | 
 |   return demuxer_.OnRtpPacket(packet); | 
 | } | 
 |  | 
 | void RtpStreamReceiverController::OnRecoveredPacket( | 
 |     const RtpPacketReceived& packet) { | 
 |   RTC_DCHECK_RUN_ON(&demuxer_sequence_); | 
 |   demuxer_.OnRtpPacket(packet); | 
 | } | 
 |  | 
 | bool RtpStreamReceiverController::AddSink(uint32_t ssrc, | 
 |                                           RtpPacketSinkInterface* sink) { | 
 |   RTC_DCHECK_RUN_ON(&demuxer_sequence_); | 
 |   return demuxer_.AddSink(ssrc, sink); | 
 | } | 
 |  | 
 | bool RtpStreamReceiverController::RemoveSink( | 
 |     const RtpPacketSinkInterface* sink) { | 
 |   RTC_DCHECK_RUN_ON(&demuxer_sequence_); | 
 |   return demuxer_.RemoveSink(sink); | 
 | } | 
 |  | 
 | }  // namespace webrtc |