| /* |
| * 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 |