tg2sip/webrtc_dsp/rtc_base/criticalsection.cc

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* Copyright 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.
*/
#include "rtc_base/criticalsection.h"
#include <time.h>
#include "rtc_base/atomicops.h"
#include "rtc_base/checks.h"
#include "rtc_base/platform_thread_types.h"
#include "rtc_base/system/unused.h"
// TODO(tommi): Split this file up to per-platform implementation files.
namespace rtc {
CriticalSection::CriticalSection() {
#if defined(WEBRTC_WIN)
InitializeCriticalSection(&crit_);
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
lock_queue_ = 0;
owning_thread_ = 0;
recursion_ = 0;
semaphore_ = dispatch_semaphore_create(0);
#else
pthread_mutexattr_t mutex_attribute;
pthread_mutexattr_init(&mutex_attribute);
pthread_mutexattr_settype(&mutex_attribute, PTHREAD_MUTEX_RECURSIVE);
pthread_mutex_init(&mutex_, &mutex_attribute);
pthread_mutexattr_destroy(&mutex_attribute);
#endif
CS_DEBUG_CODE(thread_ = 0);
CS_DEBUG_CODE(recursion_count_ = 0);
RTC_UNUSED(thread_);
RTC_UNUSED(recursion_count_);
#else
#error Unsupported platform.
#endif
}
CriticalSection::~CriticalSection() {
#if defined(WEBRTC_WIN)
DeleteCriticalSection(&crit_);
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
dispatch_release(semaphore_);
#else
pthread_mutex_destroy(&mutex_);
#endif
#else
#error Unsupported platform.
#endif
}
void CriticalSection::Enter() const RTC_EXCLUSIVE_LOCK_FUNCTION() {
#if defined(WEBRTC_WIN)
EnterCriticalSection(&crit_);
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
int spin = 3000;
PlatformThreadRef self = CurrentThreadRef();
bool have_lock = false;
do {
// Instead of calling TryEnter() in this loop, we do two interlocked
// operations, first a read-only one in order to avoid affecting the lock
// cache-line while spinning, in case another thread is using the lock.
if (!IsThreadRefEqual(owning_thread_, self)) {
if (AtomicOps::AcquireLoad(&lock_queue_) == 0) {
if (AtomicOps::CompareAndSwap(&lock_queue_, 0, 1) == 0) {
have_lock = true;
break;
}
}
} else {
AtomicOps::Increment(&lock_queue_);
have_lock = true;
break;
}
sched_yield();
} while (--spin);
if (!have_lock && AtomicOps::Increment(&lock_queue_) > 1) {
// Owning thread cannot be the current thread since TryEnter() would
// have succeeded.
RTC_DCHECK(!IsThreadRefEqual(owning_thread_, self));
// Wait for the lock to become available.
dispatch_semaphore_wait(semaphore_, DISPATCH_TIME_FOREVER);
RTC_DCHECK(owning_thread_ == 0);
RTC_DCHECK(!recursion_);
}
owning_thread_ = self;
++recursion_;
#else
pthread_mutex_lock(&mutex_);
#endif
#if CS_DEBUG_CHECKS
if (!recursion_count_) {
RTC_DCHECK(!thread_);
thread_ = CurrentThreadRef();
} else {
RTC_DCHECK(CurrentThreadIsOwner());
}
++recursion_count_;
#endif
#else
#error Unsupported platform.
#endif
}
bool CriticalSection::TryEnter() const RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true) {
#if defined(WEBRTC_WIN)
return TryEnterCriticalSection(&crit_) != FALSE;
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
if (!IsThreadRefEqual(owning_thread_, CurrentThreadRef())) {
if (AtomicOps::CompareAndSwap(&lock_queue_, 0, 1) != 0)
return false;
owning_thread_ = CurrentThreadRef();
RTC_DCHECK(!recursion_);
} else {
AtomicOps::Increment(&lock_queue_);
}
++recursion_;
#else
if (pthread_mutex_trylock(&mutex_) != 0)
return false;
#endif
#if CS_DEBUG_CHECKS
if (!recursion_count_) {
RTC_DCHECK(!thread_);
thread_ = CurrentThreadRef();
} else {
RTC_DCHECK(CurrentThreadIsOwner());
}
++recursion_count_;
#endif
return true;
#else
#error Unsupported platform.
#endif
}
void CriticalSection::Leave() const RTC_UNLOCK_FUNCTION() {
RTC_DCHECK(CurrentThreadIsOwner());
#if defined(WEBRTC_WIN)
LeaveCriticalSection(&crit_);
#elif defined(WEBRTC_POSIX)
#if CS_DEBUG_CHECKS
--recursion_count_;
RTC_DCHECK(recursion_count_ >= 0);
if (!recursion_count_)
thread_ = 0;
#endif
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
RTC_DCHECK(IsThreadRefEqual(owning_thread_, CurrentThreadRef()));
RTC_DCHECK_GE(recursion_, 0);
--recursion_;
if (!recursion_)
owning_thread_ = 0;
if (AtomicOps::Decrement(&lock_queue_) > 0 && !recursion_)
dispatch_semaphore_signal(semaphore_);
#else
pthread_mutex_unlock(&mutex_);
#endif
#else
#error Unsupported platform.
#endif
}
bool CriticalSection::CurrentThreadIsOwner() const {
#if defined(WEBRTC_WIN)
// OwningThread has type HANDLE but actually contains the Thread ID:
// http://stackoverflow.com/questions/12675301/why-is-the-owningthread-member-of-critical-section-of-type-handle-when-it-is-de
// Converting through size_t avoids the VS 2015 warning C4312: conversion from
// 'type1' to 'type2' of greater size
return crit_.OwningThread ==
reinterpret_cast<HANDLE>(static_cast<size_t>(GetCurrentThreadId()));
#elif defined(WEBRTC_POSIX)
#if CS_DEBUG_CHECKS
return IsThreadRefEqual(thread_, CurrentThreadRef());
#else
return true;
#endif // CS_DEBUG_CHECKS
#else
#error Unsupported platform.
#endif
}
CritScope::CritScope(const CriticalSection* cs) : cs_(cs) {
cs_->Enter();
}
CritScope::~CritScope() {
cs_->Leave();
}
TryCritScope::TryCritScope(const CriticalSection* cs)
: cs_(cs), locked_(cs->TryEnter()) {
CS_DEBUG_CODE(lock_was_called_ = false);
RTC_UNUSED(lock_was_called_);
}
TryCritScope::~TryCritScope() {
CS_DEBUG_CODE(RTC_DCHECK(lock_was_called_));
if (locked_)
cs_->Leave();
}
bool TryCritScope::locked() const {
CS_DEBUG_CODE(lock_was_called_ = true);
return locked_;
}
void GlobalLockPod::Lock() {
#if !defined(WEBRTC_WIN) && (!defined(WEBRTC_MAC) || USE_NATIVE_MUTEX_ON_MAC)
const struct timespec ts_null = {0};
#endif
while (AtomicOps::CompareAndSwap(&lock_acquired, 0, 1)) {
#if defined(WEBRTC_WIN)
::Sleep(0);
#elif defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
sched_yield();
#else
nanosleep(&ts_null, nullptr);
#endif
}
}
void GlobalLockPod::Unlock() {
int old_value = AtomicOps::CompareAndSwap(&lock_acquired, 1, 0);
RTC_DCHECK_EQ(1, old_value) << "Unlock called without calling Lock first";
}
GlobalLock::GlobalLock() {
lock_acquired = 0;
}
GlobalLockScope::GlobalLockScope(GlobalLockPod* lock) : lock_(lock) {
lock_->Lock();
}
GlobalLockScope::~GlobalLockScope() {
lock_->Unlock();
}
} // namespace rtc