/* * 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 #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 RtpStreamReceiverController::CreateReceiver(uint32_t ssrc, RtpPacketSinkInterface* sink) { return std::make_unique(this, ssrc, sink); } bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) { RTC_DCHECK_RUN_ON(&demuxer_sequence_); return 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