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.
49 lines
1.5 KiB
C++
49 lines
1.5 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.h.
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#ifndef RTC_BASE_THREAD_CHECKER_IMPL_H_
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#define RTC_BASE_THREAD_CHECKER_IMPL_H_
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/platform_thread_types.h"
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namespace rtc {
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// Real implementation of ThreadChecker, for use in debug mode, or
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// for temporary use in release mode (e.g. to RTC_CHECK on a threading issue
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// seen only in the wild).
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//
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// Note: You should almost always use the ThreadChecker class to get the
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// right version for your build configuration.
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class ThreadCheckerImpl {
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public:
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ThreadCheckerImpl();
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~ThreadCheckerImpl();
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bool CalledOnValidThread() const;
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// Changes the thread that is checked for in CalledOnValidThread. This may
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// be useful when an object may be created on one thread and then used
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// exclusively on another thread.
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void DetachFromThread();
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private:
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CriticalSection lock_;
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// This is mutable so that CalledOnValidThread can set it.
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// It's guarded by |lock_|.
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mutable PlatformThreadRef valid_thread_;
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};
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} // namespace rtc
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#endif // RTC_BASE_THREAD_CHECKER_IMPL_H_
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