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b2b2031457
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
71 lines
2.3 KiB
C++
71 lines
2.3 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 <memory>
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#include "rtc_base/logging.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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RTC_LOG(LS_ERROR)
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<< "RtpStreamReceiverController::Receiver::Receiver: Sink "
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"could not be added for SSRC="
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<< 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() {
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// At this level the demuxer is only configured to demux by SSRC, so don't
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// worry about MIDs (MIDs are handled by upper layers).
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demuxer_.set_use_mid(false);
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}
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RtpStreamReceiverController::~RtpStreamReceiverController() = default;
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std::unique_ptr<RtpStreamReceiverInterface>
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RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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return std::make_unique<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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