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/*
* 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 <memory>
#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
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