| /* | |
| * 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 |