tg2sip/libtgvoip/webrtc_dsp/rtc_base/criticalsection.h

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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 2004 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_CRITICALSECTION_H_
#define RTC_BASE_CRITICALSECTION_H_
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/platform_thread_types.h"
#include "rtc_base/thread_annotations.h"
#if defined(WEBRTC_WIN)
// clang-format off
// clang formating would change include order.
// Include winsock2.h before including <windows.h> to maintain consistency with
// win32.h. To include win32.h directly, it must be broken out into its own
// build target.
#include <winsock2.h>
#include <windows.h>
#include <sal.h> // must come after windows headers.
// clang-format on
#endif // defined(WEBRTC_WIN)
#if defined(WEBRTC_POSIX)
#include <pthread.h>
#endif
// See notes in the 'Performance' unit test for the effects of this flag.
#define USE_NATIVE_MUTEX_ON_MAC 0
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
#include <dispatch/dispatch.h>
#endif
#define CS_DEBUG_CHECKS RTC_DCHECK_IS_ON
#if CS_DEBUG_CHECKS
#define CS_DEBUG_CODE(x) x
#else // !CS_DEBUG_CHECKS
#define CS_DEBUG_CODE(x)
#endif // !CS_DEBUG_CHECKS
namespace rtc {
// Locking methods (Enter, TryEnter, Leave)are const to permit protecting
// members inside a const context without requiring mutable CriticalSections
// everywhere.
class RTC_LOCKABLE CriticalSection {
public:
CriticalSection();
~CriticalSection();
void Enter() const RTC_EXCLUSIVE_LOCK_FUNCTION();
bool TryEnter() const RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true);
void Leave() const RTC_UNLOCK_FUNCTION();
private:
// Use only for RTC_DCHECKing.
bool CurrentThreadIsOwner() const;
#if defined(WEBRTC_WIN)
mutable CRITICAL_SECTION crit_;
#elif defined(WEBRTC_POSIX)
#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
// Number of times the lock has been locked + number of threads waiting.
// TODO(tommi): We could use this number and subtract the recursion count
// to find places where we have multiple threads contending on the same lock.
mutable volatile int lock_queue_;
// |recursion_| represents the recursion count + 1 for the thread that owns
// the lock. Only modified by the thread that owns the lock.
mutable int recursion_;
// Used to signal a single waiting thread when the lock becomes available.
mutable dispatch_semaphore_t semaphore_;
// The thread that currently holds the lock. Required to handle recursion.
mutable PlatformThreadRef owning_thread_;
#else
mutable pthread_mutex_t mutex_;
#endif
mutable PlatformThreadRef thread_; // Only used by RTC_DCHECKs.
mutable int recursion_count_; // Only used by RTC_DCHECKs.
#else // !defined(WEBRTC_WIN) && !defined(WEBRTC_POSIX)
#error Unsupported platform.
#endif
};
// CritScope, for serializing execution through a scope.
class RTC_SCOPED_LOCKABLE CritScope {
public:
explicit CritScope(const CriticalSection* cs) RTC_EXCLUSIVE_LOCK_FUNCTION(cs);
~CritScope() RTC_UNLOCK_FUNCTION();
private:
const CriticalSection* const cs_;
RTC_DISALLOW_COPY_AND_ASSIGN(CritScope);
};
// Tries to lock a critical section on construction via
// CriticalSection::TryEnter, and unlocks on destruction if the
// lock was taken. Never blocks.
//
// IMPORTANT: Unlike CritScope, the lock may not be owned by this thread in
// subsequent code. Users *must* check locked() to determine if the
// lock was taken. If you're not calling locked(), you're doing it wrong!
class TryCritScope {
public:
explicit TryCritScope(const CriticalSection* cs);
~TryCritScope();
#if defined(WEBRTC_WIN)
_Check_return_ bool locked() const;
#elif defined(WEBRTC_POSIX)
bool locked() const __attribute__((__warn_unused_result__));
#else // !defined(WEBRTC_WIN) && !defined(WEBRTC_POSIX)
#error Unsupported platform.
#endif
private:
const CriticalSection* const cs_;
const bool locked_;
mutable bool lock_was_called_; // Only used by RTC_DCHECKs.
RTC_DISALLOW_COPY_AND_ASSIGN(TryCritScope);
};
// A POD lock used to protect global variables. Do NOT use for other purposes.
// No custom constructor or private data member should be added.
class RTC_LOCKABLE GlobalLockPod {
public:
void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION();
void Unlock() RTC_UNLOCK_FUNCTION();
volatile int lock_acquired;
};
class GlobalLock : public GlobalLockPod {
public:
GlobalLock();
};
// GlobalLockScope, for serializing execution through a scope.
class RTC_SCOPED_LOCKABLE GlobalLockScope {
public:
explicit GlobalLockScope(GlobalLockPod* lock)
RTC_EXCLUSIVE_LOCK_FUNCTION(lock);
~GlobalLockScope() RTC_UNLOCK_FUNCTION();
private:
GlobalLockPod* const lock_;
RTC_DISALLOW_COPY_AND_ASSIGN(GlobalLockScope);
};
} // namespace rtc
#endif // RTC_BASE_CRITICALSECTION_H_