C++ std::recursive_mutex类代码示例

void processFdbEntriesForAging()
{
    SWSS_LOG_ENTER();

    if (!g_recursive_mutex.try_lock())
    {
        return;
    }

    SWSS_LOG_INFO("fdb infos to process: %zu", g_fdb_info_set.size());

    uint32_t current = (uint32_t)time(NULL);

    // find aged fdb entries

    for (auto it = g_fdb_info_set.begin(); it != g_fdb_info_set.end();)
    {
        sai_attribute_t attr;

        attr.id = SAI_SWITCH_ATTR_FDB_AGING_TIME;

        sai_status_t status = vs_generic_get(SAI_OBJECT_TYPE_SWITCH, it->fdb_entry.switch_id, 1, &attr);

        if (status != SAI_STATUS_SUCCESS)
        {
            SWSS_LOG_WARN("failed to get FDB aging time for switch %s",
                    sai_serialize_object_id(it->fdb_entry.switch_id).c_str());

            ++it;
            continue;
        }

        uint32_t aging_time = attr.value.u32;

        if (aging_time == 0)
        {
            // aging is disabled
            ++it;
            continue;
        }

        if ((current - it->timestamp) >= aging_time)
        {
            fdb_info_t fi = *it;

            processFdbInfo(fi, SAI_FDB_EVENT_AGED);

            it = g_fdb_info_set.erase(it);
        }
        else
        {
            ++it;
        }
    }

    g_recursive_mutex.unlock();
}

void FifoRecorder::StartRecording(s32 numFrames, CallbackFunc finishedCb)
{
	sMutex.lock();

	delete m_File;
	delete []m_Ram;
	delete []m_ExRam;

	m_File = new FifoDataFile;

	m_Ram = new u8[Memory::RAM_SIZE];
	m_ExRam = new u8[Memory::EXRAM_SIZE];
	memset(m_Ram, 0, Memory::RAM_SIZE);
	memset(m_ExRam, 0, Memory::EXRAM_SIZE);

	m_File->SetIsWii(SConfig::GetInstance().m_LocalCoreStartupParameter.bWii);

	if (!m_IsRecording)
	{
		m_WasRecording = false;
		m_IsRecording = true;
		m_RecordFramesRemaining = numFrames;
	}

	m_RequestedRecordingEnd = false;
	m_FinishedCb = finishedCb;

	sMutex.unlock();
}

// numFrames is number of stereo frames.
// This is called from *outside* the emulator thread.
int __AudioMix(short *outstereo, int numFrames)
{
	// TODO: if mixFrequency != the actual output frequency, resample!

	section.lock();
	int underrun = -1;
	s16 sampleL = 0;
	s16 sampleR = 0;
	bool anythingToPlay = false;
	for (int i = 0; i < numFrames; i++) {
		if (outAudioQueue.size() >= 2)
		{
			sampleL = outAudioQueue.pop_front();
			sampleR = outAudioQueue.pop_front();
			outstereo[i * 2 + 0] = sampleL;
			outstereo[i * 2 + 1] = sampleR;
			anythingToPlay = true;
		} else {
			if (underrun == -1) underrun = i;
			outstereo[i * 2 + 0] = sampleL;  // repeat last sample, can reduce clicking
			outstereo[i * 2 + 1] = sampleR;  // repeat last sample, can reduce clicking
		}
	}
	if (anythingToPlay && underrun >= 0) {
		DEBUG_LOG(HLE, "Audio out buffer UNDERRUN at %i of %i", underrun, numFrames);
	} else {
		// DEBUG_LOG(HLE, "No underrun, mixed %i samples fine", numFrames);
	}
	section.unlock();
	return underrun >= 0 ? underrun : numFrames;
}

void __AudioDoState(PointerWrap &p)
{
	section.lock();

	p.Do(eventAudioUpdate);
	CoreTiming::RestoreRegisterEvent(eventAudioUpdate, "AudioUpdate", &hleAudioUpdate);
	p.Do(eventHostAudioUpdate);
	CoreTiming::RestoreRegisterEvent(eventHostAudioUpdate, "AudioUpdateHost", &hleHostAudioUpdate);

	p.Do(mixFrequency);
	outAudioQueue.DoState(p);

	int chanCount = ARRAY_SIZE(chans);
	p.Do(chanCount);
	if (chanCount != ARRAY_SIZE(chans))
	{
		ERROR_LOG(HLE, "Savestate failure: different number of audio channels.");
		section.unlock();
		return;
	}
	for (int i = 0; i < chanCount; ++i)
		chans[i].DoState(p);

	section.unlock();
	p.DoMarker("sceAudio");
}

u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking)
{
	u32 ret = 0;
	section.lock();
	if (chan.sampleAddress == 0)
		return SCE_ERROR_AUDIO_NOT_OUTPUT;
	if (chan.sampleQueue.size() > chan.sampleCount*2*chanQueueMaxSizeFactor) {
		// Block!
		if (blocking) {
			chan.waitingThread = __KernelGetCurThread();
			// WARNING: This changes currentThread so must grab waitingThread before (line above).
			__KernelWaitCurThread(WAITTYPE_AUDIOCHANNEL, (SceUID)chanNum, 0, 0, false, "blocking audio waited");
			// Fall through to the sample queueing, don't want to lose the samples even though
			// we're getting full.
		}
		else
		{
			chan.waitingThread = 0;
			return SCE_ERROR_AUDIO_CHANNEL_BUSY;
		}
	}
	if (chan.format == PSP_AUDIO_FORMAT_STEREO)
	{
		const u32 totalSamples = chan.sampleCount * 2;

		if (IS_LITTLE_ENDIAN)
		{
			s16 *sampleData = (s16 *) Memory::GetPointer(chan.sampleAddress);

			// Walking a pointer for speed.  But let's make sure we wouldn't trip on an invalid ptr.
			if (Memory::IsValidAddress(chan.sampleAddress + (totalSamples - 1) * sizeof(s16)))
			{
				for (u32 i = 0; i < totalSamples; i++)
					chan.sampleQueue.push(*sampleData++);
			}
		}
		else
		{
			for (u32 i = 0; i < totalSamples; i++)
				chan.sampleQueue.push((s16)Memory::Read_U16(chan.sampleAddress + sizeof(s16) * i));
		}

		ret = chan.sampleCount;
	}
	else if (chan.format == PSP_AUDIO_FORMAT_MONO)
	{
		for (u32 i = 0; i < chan.sampleCount; i++)
		{
			// Expand to stereo
			s16 sample = (s16)Memory::Read_U16(chan.sampleAddress + 2 * i);
			chan.sampleQueue.push(sample);
			chan.sampleQueue.push(sample);
		}

		ret = chan.sampleCount;
	}
	section.unlock();
	return ret;
}

int main()
{
    m.lock();
    std::thread t(f);
    std::this_thread::sleep_for(ms(250));
    m.unlock();
    t.join();
}

// Mix samples from the various audio channels into a single sample queue.
// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
// just sleep the main emulator thread a little.
void __AudioUpdate()
{
	// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
	// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
	// if the buffer somehow gets full.

	s32 mixBuffer[hwBlockSize * 2];
	memset(mixBuffer, 0, sizeof(mixBuffer));

	for (u32 i = 0; i < PSP_AUDIO_CHANNEL_MAX + 1; i++)
	{
		if (!chans[i].reserved)
			continue;
		__AudioWakeThreads(chans[i], hwBlockSize);

		if (!chans[i].sampleQueue.size()) {
			// ERROR_LOG(HLE, "No queued samples, skipping channel %i", i);
			continue;
		}

		for (int s = 0; s < hwBlockSize; s++)
		{
			if (chans[i].sampleQueue.size() >= 2)
			{
				s16 sampleL = chans[i].sampleQueue.pop_front();
				s16 sampleR = chans[i].sampleQueue.pop_front();
				mixBuffer[s * 2 + 0] += sampleL;
				mixBuffer[s * 2 + 1] += sampleR;
			} 
			else
			{
				ERROR_LOG(HLE, "Channel %i buffer underrun at %i of %i", i, s, hwBlockSize);
				break;
			}
		}
	}

	if (g_Config.bEnableSound) {
		section.lock();
		if (outAudioQueue.room() >= hwBlockSize * 2) {
			// Push the mixed samples onto the output audio queue.
			for (int i = 0; i < hwBlockSize; i++) {
				s16 sampleL = clamp_s16(mixBuffer[i * 2 + 0]);
				s16 sampleR = clamp_s16(mixBuffer[i * 2 + 1]);

				outAudioQueue.push((s16)sampleL);
				outAudioQueue.push((s16)sampleR);
			}
		} else {
			// This happens quite a lot. There's still something slightly off
			// about the amount of audio we produce.
			DEBUG_LOG(HLE, "Audio outbuffer overrun! room = %i / %i", outAudioQueue.room(), (u32)outAudioQueue.capacity());
		}
		section.unlock();
	}
	
}

void f()
{
    time_point t0 = Clock::now();
    m.lock();
    time_point t1 = Clock::now();
    m.lock();
    m.unlock();
    m.unlock();
    ns d = t1 - t0 - ms(250);
    assert(d < ns(2500000));  // within 2.5ms
}

void FifoPlayerDlg::RecordingFinished()
{
	sMutex.lock();

	if (m_EvtHandler)
	{
		wxCommandEvent event(RECORDING_FINISHED_EVENT);
		m_EvtHandler->AddPendingEvent(event);
	}

	sMutex.unlock();
}

void FifoPlayerDlg::FileLoaded()
{
	sMutex.lock();

	if (m_EvtHandler)
	{
		wxPaintEvent event;
		m_EvtHandler->AddPendingEvent(event);
	}

	sMutex.unlock();
}

void FifoPlayerDlg::FrameWritten()
{
	sMutex.lock();

	if (m_EvtHandler)
	{
		wxCommandEvent event(FRAME_WRITTEN_EVENT);
		m_EvtHandler->AddPendingEvent(event);
	}

	sMutex.unlock();
}

void attemp_10k_increasesR() {
	int i;
	mrtx.lock();
	mrtx.lock(); // 递归锁，多次调用lock都不会死锁
	for (i=0; i<10; ++i) {
		++counter;
		std::this_thread::sleep_for(std::chrono::microseconds(1*1000000));
		std::cout << "thread Reverse[" << std::this_thread::get_id() <<"]" << "add the counter" << endl; 

	}

	mrtx.unlock();
}

FifoPlayerDlg::FifoPlayerDlg(wxWindow * const parent) :
	wxDialog(parent, wxID_ANY, _("FIFO Player"), wxDefaultPosition, wxDefaultSize),
	m_FramesToRecord(1)
{
	CreateGUIControls();

	sMutex.lock();	
	m_EvtHandler = GetEventHandler();	
	sMutex.unlock();

	FifoPlayer::GetInstance().SetFileLoadedCallback(FileLoaded);
	FifoPlayer::GetInstance().SetFrameWrittenCallback(FrameWritten);
}

// Read the Wiimote once
void Update(int _WiimoteNumber)
{
	// Try to get a lock and return without doing anything if we fail
	// This avoids deadlocks when adding a Wiimote during continuous scan
	if(!g_refresh_lock.try_lock())
		return;

	if (g_wiimotes[_WiimoteNumber])
		g_wiimotes[_WiimoteNumber]->Update();

	// Wiimote::Update() may remove the Wiimote if it was disconnected.
	if (!g_wiimotes[_WiimoteNumber])
	{
		Host_ConnectWiimote(_WiimoteNumber, false);
	}
	g_refresh_lock.unlock();
}

// Mix samples from the various audio channels into a single sample queue.
// This single sample queue is where __AudioMix should read from. If the sample queue is full, we should
// just sleep the main emulator thread a little.
void __AudioUpdate()
{
	// Audio throttle doesn't really work on the PSP since the mixing intervals are so closely tied
	// to the CPU. Much better to throttle the frame rate on frame display and just throw away audio
	// if the buffer somehow gets full.

	s32 mixBuffer[hwBlockSize * 2];
	memset(mixBuffer, 0, sizeof(mixBuffer));

	for (int i = 0; i < PSP_AUDIO_CHANNEL_MAX; i++)
	{
		if (!chans[i].reserved)
			continue;
		if (!chans[i].sampleQueue.size()) {
			// ERROR_LOG(HLE, "No queued samples, skipping channel %i", i);
			continue;
		}

		for (int s = 0; s < hwBlockSize; s++)
		{
			if (chans[i].sampleQueue.size() >= 2)
			{
				s16 sampleL = chans[i].sampleQueue.pop_front();
				s16 sampleR = chans[i].sampleQueue.pop_front();
				mixBuffer[s * 2 + 0] += sampleL;
				mixBuffer[s * 2 + 1] += sampleR;
			} 
			else
			{
				ERROR_LOG(HLE, "Channel %i buffer underrun at %i of %i", i, s, hwBlockSize);
				break;
			}
		}

		if (chans[i].sampleQueue.size() < chans[i].sampleCount * 2 * chanQueueMinSizeFactor)
		{
			// Ask the thread to send more samples until next time, queue is being drained.
			if (chans[i].waitingThread) {
				SceUID waitingThread = chans[i].waitingThread;
				chans[i].waitingThread = 0;
				// DEBUG_LOG(HLE, "Woke thread %i for some buffer filling", waitingThread);
				__KernelResumeThreadFromWait(waitingThread, chans[i].sampleCount);
			}
		}
	}

	if (g_Config.bEnableSound) {
		section.lock();
		if (outAudioQueue.room() >= hwBlockSize * 2) {
			// Push the mixed samples onto the output audio queue.
			for (int i = 0; i < hwBlockSize; i++) {
				s32 sampleL = mixBuffer[i * 2 + 0] >> 2;  // TODO - what factor?
				s32 sampleR = mixBuffer[i * 2 + 1] >> 2;

				outAudioQueue.push((s16)sampleL);
				outAudioQueue.push((s16)sampleR);
			}
		} else {