| /* |
| * 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 "webrtc/call/rtp_stream_receiver_controller.h" |
| |
| #include "webrtc/rtc_base/logging.h" |
| #include "webrtc/rtc_base/ptr_util.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) { |
| LOG(LS_ERROR) << "RtpStreamReceiverController::Receiver::Receiver: Sink " |
| << "could not be added for SSRC=" << ssrc << "."; |
| } |
| } |
| |
| RtpStreamReceiverController::Receiver::~Receiver() { |
| // Don't require return value > 0, since for RTX we currently may |
| // have multiple Receiver objects with the same sink. |
| // TODO(nisse): Consider adding a DCHECK when RtxReceiveStream is wired up. |
| controller_->RemoveSink(sink_); |
| } |
| |
| RtpStreamReceiverController::RtpStreamReceiverController() = default; |
| RtpStreamReceiverController::~RtpStreamReceiverController() = default; |
| |
| std::unique_ptr<RtpStreamReceiverInterface> |
| RtpStreamReceiverController::CreateReceiver( |
| uint32_t ssrc, |
| RtpPacketSinkInterface* sink) { |
| return rtc::MakeUnique<Receiver>(this, ssrc, sink); |
| } |
| |
| bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) { |
| rtc::CritScope cs(&lock_); |
| return demuxer_.OnRtpPacket(packet); |
| } |
| |
| bool RtpStreamReceiverController::AddSink(uint32_t ssrc, |
| RtpPacketSinkInterface* sink) { |
| rtc::CritScope cs(&lock_); |
| return demuxer_.AddSink(ssrc, sink); |
| } |
| |
| size_t RtpStreamReceiverController::RemoveSink( |
| const RtpPacketSinkInterface* sink) { |
| rtc::CritScope cs(&lock_); |
| return demuxer_.RemoveSink(sink); |
| } |
| |
| } // namespace webrtc |