webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00

56 lines
1.6 KiB
C++

/*
* Copyright (c) 2015 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// An IsacBandwidthInfo that's safe to access from multiple threads because
// it's protected by a mutex.
class LockedIsacBandwidthInfo final {
public:
LockedIsacBandwidthInfo();
~LockedIsacBandwidthInfo();
IsacBandwidthInfo Get() const {
rtc::CritScope lock(&lock_);
return bwinfo_;
}
void Set(const IsacBandwidthInfo& bwinfo) {
rtc::CritScope lock(&lock_);
bwinfo_ = bwinfo;
}
int AddRef() const { return rtc::AtomicOps::Increment(&ref_count_); }
int Release() const {
const int count = rtc::AtomicOps::Decrement(&ref_count_);
if (count == 0) {
delete this;
}
return count;
}
private:
mutable volatile int ref_count_;
rtc::CriticalSection lock_;
IsacBandwidthInfo bwinfo_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_