webrtc/call/rtp_stream_receiver_controller.cc
Tomas Gunnarsson e091fd21d6 Remove lock from RtpStreamReceiverController.
The demuxer variable is now being used from the same thread consistently
so it's safe to replace the lock with a sequence checker.

Down the line, we may move construction+use of the
RtpStreamReceiverController class in Call, over to the network thread.
This should be possible without further modifications to
RtpStreamReceiverController.

Bug: webrtc:11993, webrtc:11567
Change-Id: Iee8c31ddf9b26b39393f40b5b1d25343b0233ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202245
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33016}
2021-01-18 09:10:14 +00:00

67 lines
2.2 KiB
C++

/*
* 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() {
// 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() {}
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);
}
bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
RtpPacketSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&demuxer_sequence_);
return demuxer_.AddSink(ssrc, sink);
}
size_t RtpStreamReceiverController::RemoveSink(
const RtpPacketSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&demuxer_sequence_);
return demuxer_.RemoveSink(sink);
}
} // namespace webrtc