mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
35 lines
1.1 KiB
C++
35 lines
1.1 KiB
C++
/*
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* Copyright (c) 2014 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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// Borrowed from Chromium's src/base/threading/thread_checker_impl.cc.
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#include "rtc_base/thread_checker_impl.h"
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namespace rtc {
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ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadRef()) {}
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ThreadCheckerImpl::~ThreadCheckerImpl() {}
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bool ThreadCheckerImpl::CalledOnValidThread() const {
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const PlatformThreadRef current_thread = CurrentThreadRef();
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CritScope scoped_lock(&lock_);
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if (!valid_thread_) // Set if previously detached.
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valid_thread_ = current_thread;
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return IsThreadRefEqual(valid_thread_, current_thread);
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}
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void ThreadCheckerImpl::DetachFromThread() {
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CritScope scoped_lock(&lock_);
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valid_thread_ = 0;
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}
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} // namespace rtc
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