webrtc/rtc_base/synchronization/mutex.h
Markus Handell f70fbc8411 Introduces rtc_base/synchronization/mutex.h.
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.

The mutex types supportable by webrtc::Mutex are

- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)

In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.

The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.

Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.

Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
2020-06-04 09:55:12 +00:00

103 lines
2.9 KiB
C++

/*
* Copyright 2020 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 RTC_BASE_SYNCHRONIZATION_MUTEX_H_
#define RTC_BASE_SYNCHRONIZATION_MUTEX_H_
#include <atomic>
#include "absl/base/const_init.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/unused.h"
#include "rtc_base/thread_annotations.h"
#if defined(WEBRTC_ABSL_MUTEX)
#include "rtc_base/synchronization/mutex_abseil.h"
#elif defined(WEBRTC_WIN)
#include "rtc_base/synchronization/mutex_critical_section.h"
#elif defined(WEBRTC_POSIX)
#include "rtc_base/synchronization/mutex_pthread.h"
#else
#error Unsupported platform.
#endif
namespace webrtc {
// The Mutex guarantees exclusive access and aims to follow Abseil semantics
// (i.e. non-reentrant etc).
class RTC_LOCKABLE Mutex final {
public:
Mutex() = default;
Mutex(const Mutex&) = delete;
Mutex& operator=(const Mutex&) = delete;
void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION() { impl_.Lock(); }
RTC_WARN_UNUSED_RESULT bool TryLock() RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true) {
return impl_.TryLock();
}
void Unlock() RTC_UNLOCK_FUNCTION() { impl_.Unlock(); }
private:
MutexImpl impl_;
};
// MutexLock, for serializing execution through a scope.
class RTC_SCOPED_LOCKABLE MutexLock final {
public:
MutexLock(const MutexLock&) = delete;
MutexLock& operator=(const MutexLock&) = delete;
explicit MutexLock(Mutex* mutex) RTC_EXCLUSIVE_LOCK_FUNCTION(mutex)
: mutex_(mutex) {
mutex->Lock();
}
~MutexLock() RTC_UNLOCK_FUNCTION() { mutex_->Unlock(); }
private:
Mutex* mutex_;
};
// A mutex used to protect global variables. Do NOT use for other purposes.
#if defined(WEBRTC_ABSL_MUTEX)
using GlobalMutex = absl::Mutex;
using GlobalMutexLock = absl::MutexLock;
#else
class RTC_LOCKABLE GlobalMutex final {
public:
GlobalMutex(const GlobalMutex&) = delete;
GlobalMutex& operator=(const GlobalMutex&) = delete;
constexpr explicit GlobalMutex(absl::ConstInitType /*unused*/)
: mutex_locked_(0) {}
void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION();
void Unlock() RTC_UNLOCK_FUNCTION();
private:
std::atomic<int> mutex_locked_; // 0 means lock not taken, 1 means taken.
};
// GlobalMutexLock, for serializing execution through a scope.
class RTC_SCOPED_LOCKABLE GlobalMutexLock final {
public:
GlobalMutexLock(const GlobalMutexLock&) = delete;
GlobalMutexLock& operator=(const GlobalMutexLock&) = delete;
explicit GlobalMutexLock(GlobalMutex* mutex) RTC_EXCLUSIVE_LOCK_FUNCTION();
~GlobalMutexLock() RTC_UNLOCK_FUNCTION();
private:
GlobalMutex* mutex_;
};
#endif // if defined(WEBRTC_ABSL_MUTEX)
} // namespace webrtc
#endif // RTC_BASE_SYNCHRONIZATION_MUTEX_H_