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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/rtp_stream_receiver_controller.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ptr_util.h"
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namespace webrtc {
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RtpStreamReceiverController::Receiver::Receiver(
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RtpStreamReceiverController* controller,
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uint32_t ssrc,
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RtpPacketSinkInterface* sink)
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: controller_(controller), sink_(sink) {
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const bool sink_added = controller_->AddSink(ssrc, sink_);
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if (!sink_added) {
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LOG(LS_ERROR) << "RtpStreamReceiverController::Receiver::Receiver: Sink "
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<< "could not be added for SSRC=" << ssrc << ".";
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}
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}
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RtpStreamReceiverController::Receiver::~Receiver() {
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// Don't require return value > 0, since for RTX we currently may
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// have multiple Receiver objects with the same sink.
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// TODO(nisse): Consider adding a DCHECK when RtxReceiveStream is wired up.
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controller_->RemoveSink(sink_);
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}
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RtpStreamReceiverController::RtpStreamReceiverController() = default;
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RtpStreamReceiverController::~RtpStreamReceiverController() = default;
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std::unique_ptr<RtpStreamReceiverInterface>
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RtpStreamReceiverController::CreateReceiver(
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uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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return rtc::MakeUnique<Receiver>(this, ssrc, sink);
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}
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bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
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rtc::CritScope cs(&lock_);
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return demuxer_.OnRtpPacket(packet);
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}
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bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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rtc::CritScope cs(&lock_);
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return demuxer_.AddSink(ssrc, sink);
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}
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size_t RtpStreamReceiverController::RemoveSink(
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const RtpPacketSinkInterface* sink) {
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rtc::CritScope cs(&lock_);
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return demuxer_.RemoveSink(sink);
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}
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} // namespace webrtc
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