HRESULT CMpcAudioRenderer::InitCoopLevel()
{
HRESULT hr = S_OK;
IVideoWindow* pVideoWindow = NULL;
HWND hWnd = NULL;
CComBSTR bstrCaption;
hr = m_pGraph->QueryInterface (__uuidof(IVideoWindow), (void**) &pVideoWindow);
if (SUCCEEDED (hr))
{
pVideoWindow->get_Owner((long*)&hWnd);
SAFE_RELEASE (pVideoWindow);
}
if (!hWnd)
{
hWnd = GetTopWindow(NULL);
}
ATLASSERT(hWnd != NULL);
if (!useWASAPI)
hr = m_pDS->SetCooperativeLevel(hWnd, DSSCL_PRIORITY);
else if (hTask == NULL)
{
// Ask MMCSS to temporarily boost the thread priority
// to reduce glitches while the low-latency stream plays.
DWORD taskIndex = 0;
hTask = AvSetMmThreadCharacteristics(TEXT("Pro Audio"), &taskIndex);
hr=GetLastError();
if (hTask == NULL)
return hr;
}
return hr;
}
void
WWThreadCharacteristics::Setup(void)
{
HRESULT hr = S_OK;
if (WWSTTNone != m_schedulerTaskType) {
// マルチメディアクラススケジューラーサービスのスレッド特性設定。
dprintf("D: %s() AvSetMmThreadCharacteristics(%S)\n",
__FUNCTION__, WWSchedulerTaskTypeToStr(m_schedulerTaskType));
m_mmcssHandle =
AvSetMmThreadCharacteristics(WWSchedulerTaskTypeToStr(m_schedulerTaskType), &m_mmcssTaskIndex);
if (nullptr == m_mmcssHandle) {
dprintf("Failed to enable MMCSS on render thread: 0x%08x\n", GetLastError());
m_mmcssTaskIndex = 0;
m_result.avSetMmThreadCharacteristicsResult = false;
} else {
m_result.avSetMmThreadCharacteristicsResult = true;
}
if (m_result.avSetMmThreadCharacteristicsResult && WWTPNone != m_threadPriority) {
// スレッド優先度設定。
dprintf("D: %s() AvSetMmThreadPriority(%S)\n",
__FUNCTION__, WWMMThreadPriorityTypeToStr(m_threadPriority));
assert(m_mmcssHandle != nullptr);
m_result.avSetMmThreadPriorityResult =
!!AvSetMmThreadPriority(m_mmcssHandle, WWMMThreadPriorityTypeToAvrtPriority(m_threadPriority));
}
}
if (WWMMCSSDoNotCall != m_mmcssCallType) {
// MMCSSの有効、無効の設定。
hr = DwmEnableMMCSS(m_mmcssCallType==WWMMCSSEnable);
dprintf("D: %s() DwmEnableMMCSS(%d) 0x%08x\n", __FUNCTION__, (int)(m_mmcssCallType==WWMMCSSEnable), hr);
// 失敗することがあるが、続行する。
m_result.dwmEnableMMCSSResult = hr;
}
}
void propagateWithRawCurrentFormat(WAVEFORMATEX *toThis) {
WAVEFORMATEX *pwfx;
IMMDevice *pMMDevice;
IAudioClient *pAudioClient;
HANDLE hTask;
DWORD nTaskIndex = 0;
hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
HRESULT hr = get_default_device(&pMMDevice);
if (FAILED(hr)) {
assert(false);
}
// activate an (the default, for us, since we want loopback) IAudioClient
hr = pMMDevice->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL, NULL,
(void**)&pAudioClient
);
if (FAILED(hr)) {
ShowOutput("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
assert(false);
}
hr = pAudioClient->GetMixFormat(&pwfx);
if (FAILED(hr)) {
ShowOutput("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
assert(false);
}
pAudioClient->Stop();
AvRevertMmThreadCharacteristics(hTask);
pAudioClient->Release();
pMMDevice->Release();
memcpy(toThis, pwfx, sizeof(WAVEFORMATEX));
CoTaskMemFree(pwfx);
}
DWORD CWASAPICapture::DoCaptureThread()
{
bool stillPlaying = true;
HANDLE waitArray[2] = {_ShutdownEvent, _StreamSwitchEvent};
HANDLE mmcssHandle = NULL;
DWORD mmcssTaskIndex = 0;
HRESULT hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
PersistentAssert(SUCCEEDED(hr), "CoInitializeEx failed");
if (!DisableMMCSS)
{
mmcssHandle = AvSetMmThreadCharacteristics("Audio", &mmcssTaskIndex);
PersistentAssert(mmcssHandle != NULL, "AvSetMmThreadCharacteristics failed");
}
while (stillPlaying)
{
HRESULT hr;
//
// In Timer Driven mode, we want to wait for half the desired latency in milliseconds.
//
// That way we'll wake up half way through the processing period to pull the
// next set of samples from the engine.
//
DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, _EngineLatencyInMS / 2);
switch (waitResult)
{
case WAIT_OBJECT_0 + 0: // _ShutdownEvent
stillPlaying = false; // We're done, exit the loop.
break;
case WAIT_OBJECT_0 + 1: // _StreamSwitchEvent
PersistentSignalError("StreamSwitch event unexpected");
stillPlaying = false;
break;
case WAIT_TIMEOUT: // Timeout
//
// We need to retrieve the next buffer of samples from the audio capturer.
//
BYTE *pData;
UINT32 framesAvailable;
DWORD flags;
//
// Find out how much capture data is available. We need to make sure we don't run over the length
// of our capture buffer. We'll discard any samples that don't fit in the buffer.
//
UINT64 CaptureStartTime;
hr = _CaptureClient->GetBuffer(&pData, &framesAvailable, &flags, NULL, &CaptureStartTime);
if (SUCCEEDED(hr))
{
UINT32 framesToCopy = min(framesAvailable, static_cast<UINT32>((_CaptureBufferSize - _CurrentCaptureIndex) / _FrameSize));
const UINT BytesToCopy = framesToCopy * _FrameSize;
if (framesToCopy != 0)
{
//
// The flags on capture tell us information about the data.
//
// We only really care about the silent flag since we want to put frames of silence into the buffer
// when we receive silence. We rely on the fact that a logical bit 0 is silence for both float and int formats.
//
if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
{
//
// Fill 0s from the capture buffer to the output buffer.
//
ZeroMemory(&_CaptureBuffer[_CurrentCaptureIndex], BytesToCopy);
}
else
{
//
// Copy data from the audio engine buffer to the output buffer.
//
CopyMemory(&_CaptureBuffer[_CurrentCaptureIndex], pData, BytesToCopy);
}
//
// Bump the capture buffer pointer.
//
if(_Compressor == NULL)
{
_CurrentCaptureIndex += BytesToCopy;
}
}
hr = _CaptureClient->ReleaseBuffer(framesAvailable);
PersistentAssert(SUCCEEDED(hr), "_CaptureClient->ReleaseBuffer failed");
if(_Compressor && framesToCopy != 0)
{
_Compressor->AudioSample32Bit2Channel((float *)_CaptureBuffer, framesToCopy, CaptureStartTime);
}
}
break;
}
}
if (!DisableMMCSS)
{
AvRevertMmThreadCharacteristics(mmcssHandle);
}
CoUninitialize();
return 0;
}
void OBS::MainAudioLoop()
{
const unsigned int audioSamplesPerSec = App->GetSampleRateHz();
const unsigned int audioSampleSize = audioSamplesPerSec/100;
DWORD taskID = 0;
HANDLE hTask = AvSetMmThreadCharacteristics(TEXT("Pro Audio"), &taskID);
bufferedAudioTimes.Clear();
bPushToTalkOn = false;
micMax = desktopMax = VOL_MIN;
micPeak = desktopPeak = VOL_MIN;
UINT audioFramesSinceMeterUpdate = 0;
UINT audioFramesSinceMicMaxUpdate = 0;
UINT audioFramesSinceDesktopMaxUpdate = 0;
List<float> mixBuffer, levelsBuffer;
mixBuffer.SetSize(audioSampleSize*2);
levelsBuffer.SetSize(audioSampleSize*2);
latestAudioTime = 0;
//---------------------------------------------
// the audio loop of doom
while (true) {
OSSleep(5); //screw it, just run it every 5ms
if (!bRunning)
break;
//-----------------------------------------------
float *desktopBuffer, *micBuffer;
curDesktopVol = desktopVol * desktopBoost;
if (bUsingPushToTalk)
curMicVol = bPushToTalkOn ? micVol : 0.0f;
else
curMicVol = micVol;
curMicVol *= micBoost;
bool bDesktopMuted = (curDesktopVol < EPSILON);
bool bMicEnabled = (micAudio != NULL);
while (QueryNewAudio()) {
QWORD timestamp = bufferedAudioTimes[0];
bufferedAudioTimes.Remove(0);
zero(mixBuffer.Array(), audioSampleSize*2*sizeof(float));
zero(levelsBuffer.Array(), audioSampleSize*2*sizeof(float));
//----------------------------------------------------------------------------
// get latest sample for calculating the volume levels
float *latestDesktopBuffer = NULL, *latestMicBuffer = NULL;
desktopAudio->GetBuffer(&desktopBuffer, timestamp);
desktopAudio->GetNewestFrame(&latestDesktopBuffer);
if (micAudio != NULL) {
micAudio->GetBuffer(&micBuffer, timestamp);
micAudio->GetNewestFrame(&latestMicBuffer);
}
//----------------------------------------------------------------------------
// mix desktop samples
if (desktopBuffer)
MixAudio(mixBuffer.Array(), desktopBuffer, audioSampleSize*2, false);
if (latestDesktopBuffer)
MixAudio(levelsBuffer.Array(), latestDesktopBuffer, audioSampleSize*2, false);
//----------------------------------------------------------------------------
// get latest aux volume level samples and mix
OSEnterMutex(hAuxAudioMutex);
for (UINT i=0; i<auxAudioSources.Num(); i++) {
float *latestAuxBuffer;
if(auxAudioSources[i]->GetNewestFrame(&latestAuxBuffer))
MixAudio(levelsBuffer.Array(), latestAuxBuffer, audioSampleSize*2, false);
}
//----------------------------------------------------------------------------
// mix output aux sound samples with the desktop
for (UINT i=0; i<auxAudioSources.Num(); i++) {
float *auxBuffer;
if(auxAudioSources[i]->GetBuffer(&auxBuffer, timestamp))
MixAudio(mixBuffer.Array(), auxBuffer, audioSampleSize*2, false);
}
OSLeaveMutex(hAuxAudioMutex);
//----------------------------------------------------------------------------
// multiply samples by volume and compute RMS and max of samples
// Use 1.0f instead of curDesktopVol, since aux audio sources already have their volume set, and shouldn't be boosted anyway.
float desktopRMS = 0, micRMS = 0, desktopMx = 0, micMx = 0;
if (latestDesktopBuffer)
CalculateVolumeLevels(levelsBuffer.Array(), audioSampleSize*2, 1.0f, desktopRMS, desktopMx);
if (bMicEnabled && latestMicBuffer)
CalculateVolumeLevels(latestMicBuffer, audioSampleSize*2, curMicVol, micRMS, micMx);
//----------------------------------------------------------------------------
// convert RMS and Max of samples to dB
desktopRMS = toDB(desktopRMS);
micRMS = toDB(micRMS);
desktopMx = toDB(desktopMx);
micMx = toDB(micMx);
//----------------------------------------------------------------------------
// update max if sample max is greater or after 1 second
float maxAlpha = 0.15f;
UINT peakMeterDelayFrames = audioSamplesPerSec * 3;
if (micMx > micMax)
micMax = micMx;
else
micMax = maxAlpha * micMx + (1.0f - maxAlpha) * micMax;
if(desktopMx > desktopMax)
desktopMax = desktopMx;
else
desktopMax = maxAlpha * desktopMx + (1.0f - maxAlpha) * desktopMax;
//----------------------------------------------------------------------------
// update delayed peak meter
if (micMax > micPeak || audioFramesSinceMicMaxUpdate > peakMeterDelayFrames) {
micPeak = micMax;
audioFramesSinceMicMaxUpdate = 0;
} else {
audioFramesSinceMicMaxUpdate += audioSampleSize;
}
if (desktopMax > desktopPeak || audioFramesSinceDesktopMaxUpdate > peakMeterDelayFrames) {
desktopPeak = desktopMax;
audioFramesSinceDesktopMaxUpdate = 0;
} else {
audioFramesSinceDesktopMaxUpdate += audioSampleSize;
}
//----------------------------------------------------------------------------
// low pass the level sampling
float rmsAlpha = 0.15f;
desktopMag = rmsAlpha * desktopRMS + desktopMag * (1.0f - rmsAlpha);
micMag = rmsAlpha * micRMS + micMag * (1.0f - rmsAlpha);
//----------------------------------------------------------------------------
// update the meter about every 50ms
audioFramesSinceMeterUpdate += audioSampleSize;
if (audioFramesSinceMeterUpdate >= (audioSampleSize*5)) {
PostMessage(hwndMain, WM_COMMAND, MAKEWPARAM(ID_MICVOLUMEMETER, VOLN_METERED), 0);
audioFramesSinceMeterUpdate = 0;
}
//----------------------------------------------------------------------------
// mix mic and desktop sound
// also, it's perfectly fine to just mix into the returned buffer
if (bMicEnabled && micBuffer)
MixAudio(mixBuffer.Array(), micBuffer, audioSampleSize*2, bForceMicMono);
EncodeAudioSegment(mixBuffer.Array(), audioSampleSize, timestamp);
}
//-----------------------------------------------
if (!bRecievedFirstAudioFrame && pendingAudioFrames.Num())
bRecievedFirstAudioFrame = true;
}
desktopMag = desktopMax = desktopPeak = VOL_MIN;
micMag = micMax = micPeak = VOL_MIN;
PostMessage(hwndMain, WM_COMMAND, MAKEWPARAM(ID_MICVOLUMEMETER, VOLN_METERED), 0);
for (UINT i=0; i<pendingAudioFrames.Num(); i++)
pendingAudioFrames[i].audioData.Clear();
AvRevertMmThreadCharacteristics(hTask);
}
/// <summary>
/// Audio capture thread. Captures audio data in a loop until it is signaled to stop.
/// </summary>
/// <returns>Non-zero if thread ended successfully, zero in case of failure</returns>
DWORD WINAPI KinectAudioStream::CaptureThread()
{
HANDLE mmHandle = NULL;
DWORD mmTaskIndex = 0;
HRESULT hr = S_OK;
bool bContinue = true;
BYTE *pbOutputBuffer = NULL;
CStaticMediaBuffer outputBuffer;
DMO_OUTPUT_DATA_BUFFER OutputBufferStruct = {0};
OutputBufferStruct.pBuffer = &outputBuffer;
DWORD dwStatus = 0;
ULONG cbProduced = 0;
// Set high priority to avoid getting preempted while capturing sound
mmHandle = AvSetMmThreadCharacteristics("Audio", &mmTaskIndex);
while (bContinue)
{
if (WaitForSingleObject(m_hStopEvent, 0) == WAIT_OBJECT_0)
{
bContinue = false;
continue;
}
do
{
outputBuffer.Init(0);
OutputBufferStruct.dwStatus = 0;
hr = m_pKinectDmo->ProcessOutput(0, 1, &OutputBufferStruct, &dwStatus);
if (FAILED(hr))
{
bContinue = false;
break;
}
if (hr == S_FALSE)
{
cbProduced = 0;
}
else
{
outputBuffer.GetBufferAndLength(&pbOutputBuffer, &cbProduced);
}
// Queue audio data to be read by IStream client
if (cbProduced > 0)
{
QueueCapturedData(pbOutputBuffer, cbProduced);
}
} while (OutputBufferStruct.dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE);
Sleep(10); //sleep 10ms
}
SetEvent(m_hDataReady);
AvRevertMmThreadCharacteristics(mmHandle);
if (FAILED(hr))
{
return 0;
}
return 1;
}
// we only call this once...per hit of the play button :)
HRESULT LoopbackCaptureSetup()
{
assert(shouldStop); // duplicate starts would be odd...
shouldStop = false; // allow graphs to restart, if they so desire...
pnFrames = 0;
bool bInt16 = true; // makes it actually work, for some reason...LODO
HRESULT hr;
hr = get_default_device(&m_pMMDevice); // so it can re-place our pointer...
if (FAILED(hr)) {
return hr;
}
// tell it to not overflow one buffer's worth <sigh> not sure if this is right or not, and thus we don't "cache" or "buffer" more than that much currently...
// but a buffer size is a buffer size...hmm...as long as we keep it small though...
assert(expectedMaxBufferSize <= pBufOriginalSize);
// activate an (the default, for us, since we want loopback) IAudioClient
hr = m_pMMDevice->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL, NULL,
(void**)&pAudioClient
);
if (FAILED(hr)) {
ShowOutput("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return hr;
}
// get the default device periodicity, why? I don't know...
REFERENCE_TIME hnsDefaultDevicePeriod;
hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, NULL);
if (FAILED(hr)) {
ShowOutput("IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
// get the default device format (incoming...)
WAVEFORMATEX *pwfx; // incoming wave...
// apparently propogated by GetMixFormat...
hr = pAudioClient->GetMixFormat(&pwfx);
if (FAILED(hr)) {
ShowOutput("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return hr;
}
if (true /*bInt16*/) {
// coerce int-16 wave format
// can do this in-place since we're not changing the size of the format
// also, the engine will auto-convert from float to int for us
switch (pwfx->wFormatTag) {
case WAVE_FORMAT_IEEE_FLOAT:
assert(false);// we never get here...I hope...
pwfx->wFormatTag = WAVE_FORMAT_PCM;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
// WE GET HERE!
pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
pEx->Samples.wValidBitsPerSample = 16;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
/* scawah lodo...
if(ifNotNullThenJustSetTypeOnly) {
PWAVEFORMATEXTENSIBLE pEx2 = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(ifNotNullThenJustSetTypeOnly);
pEx2->SubFormat = pEx->SubFormat;
pEx2->Samples.wValidBitsPerSample = pEx->Samples.wValidBitsPerSample;
} */
} else {
ShowOutput("Don't know how to coerce mix format to int-16\n");
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
break;
default:
ShowOutput("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", pwfx->wFormatTag);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
/* scawah setting stream types up to match...didn't seem to work well...
if(ifNotNullThenJustSetTypeOnly) {
// pwfx is set at this point...
WAVEFORMATEX* pwfex = ifNotNullThenJustSetTypeOnly;
// copy them all out as the possible format...hmm...
pwfx->wFormatTag = WAVE_FORMAT_PCM;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
pwfex->wFormatTag = pwfx->wFormatTag;
pwfex->nChannels = pwfx->nChannels;
pwfex->nSamplesPerSec = pwfx->nSamplesPerSec;
pwfex->wBitsPerSample = pwfx->wBitsPerSample;
pwfex->nBlockAlign = pwfx->nBlockAlign;
pwfex->nAvgBytesPerSec = pwfx->nAvgBytesPerSec;
pwfex->cbSize = pwfx->cbSize;
//FILE *fp = fopen("/normal2", "w"); // fails on me? maybe juts a VLC thing...
//fShowOutput(fp, "hello world %d %d %d %d %d %d %d", pwfex->wFormatTag, pwfex->nChannels,
// pwfex->nSamplesPerSec, pwfex->wBitsPerSample, pwfex->nBlockAlign, pwfex->nAvgBytesPerSec, pwfex->cbSize );
//fclose(fp);
// cleanup
// I might be leaking here...
CoTaskMemFree(pwfx);
pAudioClient->Release();
//m_pMMDevice->Release();
return hr;
}*/
MMCKINFO ckRIFF = {0};
MMCKINFO ckData = {0};
nBlockAlign = pwfx->nBlockAlign;
// avoid stuttering on close
// http://social.msdn.microsoft.com/forums/en-US/windowspro-audiodevelopment/thread/c7ba0a04-46ce-43ff-ad15-ce8932c00171/
//IAudioClient *pAudioClient = NULL;
//IAudioCaptureClient *pCaptureClient = NULL;
IMMDeviceEnumerator *pEnumerator = NULL;
IMMDevice *pDevice = NULL;
IAudioRenderClient *pRenderClient = NULL;
WAVEFORMATEXTENSIBLE *captureDataFormat = NULL;
BYTE *captureData;
REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;
hr = CoCreateInstance(
CLSID_MMDeviceEnumerator, NULL,
CLSCTX_ALL, IID_IMMDeviceEnumerator,
(void**)&pEnumerator);
EXIT_ON_ERROR(hr)
hr = pEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &pDevice);
EXIT_ON_ERROR(hr)
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
EXIT_ON_ERROR(hr)
hr = pAudioClient->GetMixFormat((WAVEFORMATEX **)&captureDataFormat);
EXIT_ON_ERROR(hr)
// Silence: initialise in sharedmode [this is the "silence" bug overwriter, so buffer doesn't matter as much...]
hr = pAudioClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
0,
REFTIMES_PER_SEC, // buffer size a full 1.0s, though prolly doesn't matter here.
0,
pwfx,
NULL);
EXIT_ON_ERROR(hr)
// get the frame count
UINT32 bufferFrameCount;
hr = pAudioClient->GetBufferSize(&bufferFrameCount);
EXIT_ON_ERROR(hr)
// create a render client
hr = pAudioClient->GetService(IID_IAudioRenderClient, (void**)&pRenderClient);
EXIT_ON_ERROR(hr)
// get the buffer
hr = pRenderClient->GetBuffer(bufferFrameCount, &captureData);
EXIT_ON_ERROR(hr)
// release it
hr = pRenderClient->ReleaseBuffer(bufferFrameCount, AUDCLNT_BUFFERFLAGS_SILENT);
EXIT_ON_ERROR(hr)
// release the audio client
pAudioClient->Release();
EXIT_ON_ERROR(hr)
// create a new IAudioClient
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
EXIT_ON_ERROR(hr)
// -============================ now the sniffing code initialization stuff, direct from mauritius... ===================================
// call IAudioClient::Initialize
// note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
// do not work together...
// the "data ready" event never gets set
// so we're going to have to do this in a timer-driven loop...
hr = pAudioClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_LOOPBACK,
REFTIMES_PER_SEC, // buffer size a full 1.0s, seems ok VLC
0, pwfx, 0
);
if (FAILED(hr)) {
ShowOutput("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
CoTaskMemFree(pwfx);
// activate an IAudioCaptureClient
hr = pAudioClient->GetService(
__uuidof(IAudioCaptureClient),
(void**)&pAudioCaptureClient // CARE INSTANTIATION
);
if (FAILED(hr)) {
ShowOutput("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
// register with MMCSS
DWORD nTaskIndex = 0;
hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
if (NULL == hTask) {
DWORD dwErr = GetLastError();
ShowOutput("AvSetMmThreadCharacteristics failed: last error = %u\n", dwErr);
pAudioCaptureClient->Release();
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
// call IAudioClient::Start
hr = pAudioClient->Start();
if (FAILED(hr)) {
ShowOutput("IAudioClient::Start failed: hr = 0x%08x\n", hr);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
pAudioClient->Release();
return hr;
}
bFirstPacket = true;
// start the forever grabbing thread...
DWORD dwThreadID;
m_hThread = CreateThread(NULL,
0,
propagateBufferForever,
0,
0,
&dwThreadID);
if(!m_hThread)
{
DWORD dwErr = GetLastError();
return HRESULT_FROM_WIN32(dwErr);
} else {
// we...shouldn't need this...maybe?
// seems to make no difference...
hr = SetThreadPriority(m_hThread, THREAD_PRIORITY_TIME_CRITICAL);
if (FAILED(hr)) { // of course we always want to be a high prio thread, right? [we don't use much cpu...]
return hr;
}
}
return hr;
} // end LoopbackCaptureSetup
void OBS::MainAudioLoop()
{
DWORD taskID = 0;
HANDLE hTask = AvSetMmThreadCharacteristics(TEXT("Pro Audio"), &taskID);
bPushToTalkOn = false;
micMax = desktopMax = VOL_MIN;
micPeak = desktopPeak = VOL_MIN;
UINT audioFramesSinceMeterUpdate = 0;
UINT audioFramesSinceMicMaxUpdate = 0;
UINT audioFramesSinceDesktopMaxUpdate = 0;
List<float> mixedLatestDesktopSamples;
List<float> blank10msSample;
blank10msSample.SetSize(882);
QWORD lastAudioTime = 0;
while(TRUE)
{
OSSleep(5); //screw it, just run it every 5ms
if(!bRunning)
break;
//-----------------------------------------------
float *desktopBuffer, *micBuffer;
UINT desktopAudioFrames = 0, micAudioFrames = 0;
UINT latestDesktopAudioFrames = 0, latestMicAudioFrames = 0;
curDesktopVol = desktopVol * desktopBoost;
if(bUsingPushToTalk)
curMicVol = bPushToTalkOn ? micVol : 0.0f;
else
curMicVol = micVol;
curMicVol *= micBoost;
bool bDesktopMuted = (curDesktopVol < EPSILON);
bool bMicEnabled = (micAudio != NULL);
QWORD timestamp;
while(QueryNewAudio(timestamp))
{
if (!lastAudioTime)
lastAudioTime = App->GetSceneTimestamp();
if (lastAudioTime < timestamp) {
while ((lastAudioTime+=10) < timestamp)
EncodeAudioSegment(blank10msSample.Array(), 441, lastAudioTime);
}
//----------------------------------------------------------------------------
// get latest sample for calculating the volume levels
float *latestDesktopBuffer = NULL, *latestMicBuffer = NULL;
desktopAudio->GetBuffer(&desktopBuffer, &desktopAudioFrames, timestamp-10);
desktopAudio->GetNewestFrame(&latestDesktopBuffer, &latestDesktopAudioFrames);
UINT totalFloats = desktopAudioFrames*2;
if(bDesktopMuted)
{
// Clearing the desktop audio buffer before mixing in the auxiliary audio sources.
zero(desktopBuffer, sizeof(*desktopBuffer)*totalFloats);
}
if(micAudio != NULL)
{
micAudio->GetBuffer(&micBuffer, &micAudioFrames, timestamp-10);
micAudio->GetNewestFrame(&latestMicBuffer, &latestMicAudioFrames);
}
//----------------------------------------------------------------------------
// get latest aux volume level samples and mix
OSEnterMutex(hAuxAudioMutex);
mixedLatestDesktopSamples.CopyArray(latestDesktopBuffer, latestDesktopAudioFrames*2);
for(UINT i=0; i<auxAudioSources.Num(); i++)
{
float *latestAuxBuffer;
if(auxAudioSources[i]->GetNewestFrame(&latestAuxBuffer, &latestDesktopAudioFrames))
MixAudio(mixedLatestDesktopSamples.Array(), latestAuxBuffer, latestDesktopAudioFrames*2, false);
}
//----------------------------------------------------------------------------
// mix output aux sound samples with the desktop
for(UINT i=0; i<auxAudioSources.Num(); i++)
{
float *auxBuffer;
if(auxAudioSources[i]->GetBuffer(&auxBuffer, &desktopAudioFrames, timestamp-10))
MixAudio(desktopBuffer, auxBuffer, desktopAudioFrames*2, false);
}
OSLeaveMutex(hAuxAudioMutex);
//----------------------------------------------------------------------------
//UINT totalFloats = desktopAudioFrames*2;
//----------------------------------------------------------------------------
/*multiply samples by volume and compute RMS and max of samples*/
float desktopRMS = 0, micRMS = 0, desktopMx = 0, micMx = 0;
// Use 1.0f instead of curDesktopVol, since aux audio sources already have their volume set, and shouldn't be boosted anyway.
if(latestDesktopBuffer)
CalculateVolumeLevels(mixedLatestDesktopSamples.Array(), latestDesktopAudioFrames*2, 1.0f, desktopRMS, desktopMx);
if(bMicEnabled && latestMicBuffer)
CalculateVolumeLevels(latestMicBuffer, latestMicAudioFrames*2, curMicVol, micRMS, micMx);
/*convert RMS and Max of samples to dB*/
desktopRMS = toDB(desktopRMS);
micRMS = toDB(micRMS);
desktopMx = toDB(desktopMx);
micMx = toDB(micMx);
/* update max if sample max is greater or after 1 second */
float maxAlpha = 0.15f;
UINT peakMeterDelayFrames = 44100 * 3;
if(micMx > micMax)
{
micMax = micMx;
}
else
{
micMax = maxAlpha * micMx + (1.0f - maxAlpha) * micMax;
}
if(desktopMx > desktopMax)
{
desktopMax = desktopMx;
}
else
{
desktopMax = maxAlpha * desktopMx + (1.0f - maxAlpha) * desktopMax;
}
/*update delayed peak meter*/
if(micMax > micPeak || audioFramesSinceMicMaxUpdate > peakMeterDelayFrames)
{
micPeak = micMax;
audioFramesSinceMicMaxUpdate = 0;
}
else
{
audioFramesSinceMicMaxUpdate += desktopAudioFrames;
}
if(desktopMax > desktopPeak || audioFramesSinceDesktopMaxUpdate > peakMeterDelayFrames)
{
desktopPeak = desktopMax;
audioFramesSinceDesktopMaxUpdate = 0;
}
else
{
audioFramesSinceDesktopMaxUpdate += desktopAudioFrames;
}
/*low pass the level sampling*/
float rmsAlpha = 0.15f;
desktopMag = rmsAlpha * desktopRMS + desktopMag * (1.0f - rmsAlpha);
micMag = rmsAlpha * micRMS + micMag * (1.0f - rmsAlpha);
/*update the meter about every 50ms*/
audioFramesSinceMeterUpdate += desktopAudioFrames;
if(audioFramesSinceMeterUpdate >= 2205)
{
PostMessage(hwndMain, WM_COMMAND, MAKEWPARAM(ID_MICVOLUMEMETER, VOLN_METERED), 0);
audioFramesSinceMeterUpdate = 0;
}
//----------------------------------------------------------------------------
// mix mic and desktop sound, using SSE2 if available
// also, it's perfectly fine to just mix into the returned buffer
if(bMicEnabled)
MixAudio(desktopBuffer, micBuffer, totalFloats, bForceMicMono);
EncodeAudioSegment(desktopBuffer, totalFloats>>1, lastAudioTime);
}
//-----------------------------------------------
if(!bRecievedFirstAudioFrame && pendingAudioFrames.Num())
bRecievedFirstAudioFrame = true;
}
desktopMag = desktopMax = desktopPeak = VOL_MIN;
micMag = micMax = micPeak = VOL_MIN;
PostMessage(hwndMain, WM_COMMAND, MAKEWPARAM(ID_MICVOLUMEMETER, VOLN_METERED), 0);
for(UINT i=0; i<pendingAudioFrames.Num(); i++)
pendingAudioFrames[i].audioData.Clear();
AvRevertMmThreadCharacteristics(hTask);
}
void LoopbackCaptureFor(IMMDevice* mmDevice, std::string filename, int secs)
{
// open new file
MMIOINFO mi = { 0 };
// some flags cause mmioOpen write to this buffer
// but not any that we're using
std::wstring wsFilename(filename.begin(), filename.end()); // mmioOpen wants a wstring
HMMIO file = mmioOpen(const_cast<LPWSTR>(wsFilename.c_str()), &mi, MMIO_WRITE | MMIO_CREATE);
time_t startTime = time(nullptr);
// activate an IAudioClient
IAudioClient* audioClient;
HRESULT hr = mmDevice->Activate(__uuidof(IAudioClient), CLSCTX_ALL, nullptr, (void**)&audioClient);
if (FAILED(hr))
{
fprintf(stderr, "IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return;
}
// get the default device periodicity
REFERENCE_TIME hnsDefaultDevicePeriod;
hr = audioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, nullptr);
if (FAILED(hr))
{
fprintf(stderr, "IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
audioClient->Release();
return;
}
// get the default device format
WAVEFORMATEX* waveform;
hr = audioClient->GetMixFormat(&waveform);
if (FAILED(hr))
{
fprintf(stderr, "IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(waveform);
audioClient->Release();
return;
}
// coerce int-16 wave format
// can do this in-place since we're not changing the size of the format
// also, the engine will auto-convert from float to int for us
switch (waveform->wFormatTag)
{
case WAVE_FORMAT_IEEE_FLOAT:
waveform->wFormatTag = WAVE_FORMAT_PCM;
waveform->wBitsPerSample = BITS_PER_SAMPLE;
waveform->nBlockAlign = BLOCK_ALIGN;
waveform->nAvgBytesPerSec = BYTE_RATE;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(waveform);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat))
{
pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
pEx->Samples.wValidBitsPerSample = BITS_PER_SAMPLE;
waveform->wBitsPerSample = BITS_PER_SAMPLE;
waveform->nBlockAlign = waveform->nChannels * BYTE_PER_SAMPLE;
waveform->nAvgBytesPerSec = waveform->nBlockAlign * waveform->nSamplesPerSec;
}
break;
}
}
MMCKINFO ckRIFF = { 0 };
MMCKINFO ckData = { 0 };
hr = WriteWaveHeader(file, waveform, &ckRIFF, &ckData);
// create a periodic waitable timer
HANDLE hWakeUp = CreateWaitableTimer(nullptr, FALSE, nullptr);
UINT32 nBlockAlign = waveform->nBlockAlign;
// call IAudioClient::Initialize
// note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
// do not work together...
// the "data ready" event never gets set
// so we're going to do a timer-driven loop
hr = audioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_LOOPBACK, 0, 0, waveform, 0);
if (FAILED(hr))
{
fprintf(stderr, "IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
// free up waveform
CoTaskMemFree(waveform);
// activate an IAudioCaptureClient
IAudioCaptureClient* audioCaptureClient;
hr = audioClient->GetService(__uuidof(IAudioCaptureClient), (void**)&audioCaptureClient);
// register with MMCSS
DWORD nTaskIndex = 0;
HANDLE hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
if (hTask == nullptr)
{
DWORD dwErr = GetLastError();
fprintf(stderr, "AvSetMmThreadCharacteristics failed: last error = %u\n", dwErr);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
// set the waitable timer
LARGE_INTEGER liFirstFire;
liFirstFire.QuadPart = -hnsDefaultDevicePeriod / 2; // negative means relative time
LONG lTimeBetweenFires = (LONG)hnsDefaultDevicePeriod / 2 / (10 * 1000); // convert to milliseconds
if (!SetWaitableTimer(hWakeUp, &liFirstFire, lTimeBetweenFires, nullptr, nullptr, FALSE))
{
DWORD dwErr = GetLastError();
fprintf(stderr, "SetWaitableTimer failed: last error = %u\n", dwErr);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
// call IAudioClient::Start
hr = audioClient->Start();
// loopback capture loop
DWORD dwWaitResult;
UINT32 frames = 0;
for (UINT32 passes = 0; ; passes++)
{
// drain data while it is available
UINT32 nextPacketSize;
for (hr = audioCaptureClient->GetNextPacketSize(&nextPacketSize);
SUCCEEDED(hr) && nextPacketSize > 0;
hr = audioCaptureClient->GetNextPacketSize(&nextPacketSize))
{
// get the captured data
BYTE* data;
UINT32 framesToRead;
DWORD dwFlags;
hr = audioCaptureClient->GetBuffer(&data, &framesToRead, &dwFlags, nullptr, nullptr);
if (FAILED(hr))
{
fprintf(stderr, "IAudioCaptureClient::GetBuffer failed on pass %u after %u frames: hr = 0x%08x\n", passes, frames, hr);
audioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
// this type of error seems to happen often, ignore it
if (dwFlags == AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
;
else if (dwFlags != 0) {
fprintf(stderr, "IAudioCaptureClient::GetBuffer set flags to 0x%08x on pass %u after %u frames\n", dwFlags, passes, frames);
audioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
if (framesToRead == 0)
{
fprintf(stderr, "IAudioCaptureClient::GetBuffer said to read 0 frames on pass %u after %u frames\n", passes, frames);
audioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
LONG lBytesToWrite = framesToRead * nBlockAlign;
#pragma prefast(suppress: __WARNING_INCORRECT_ANNOTATION, "IAudioCaptureClient::GetBuffer SAL annotation implies a 1-byte buffer")
LONG lBytesWritten = mmioWrite(file, reinterpret_cast<PCHAR>(data), lBytesToWrite);
if (lBytesToWrite != lBytesWritten)
{
fprintf(stderr, "mmioWrite wrote %u bytes on pass %u after %u frames: expected %u bytes\n", lBytesWritten, passes, frames, lBytesToWrite);
audioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
return;
}
frames += framesToRead;
hr = audioCaptureClient->ReleaseBuffer(framesToRead);
}
dwWaitResult = WaitForSingleObject(hWakeUp, INFINITE);
if (time(nullptr) - startTime > secs)
break;
}
FinishWaveFile(file, &ckData, &ckRIFF);
audioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
audioCaptureClient->Release();
CloseHandle(hWakeUp);
audioClient->Release();
// everything went well... fixup the fact chunk in the file
MMRESULT result = mmioClose(file, 0);
file = nullptr;
if (result != MMSYSERR_NOERROR)
{
fprintf(stderr, "mmioClose failed: MMSYSERR = %u\n", result);
return;
}
// reopen the file in read/write mode
mi = { 0 };
file = mmioOpen(const_cast<LPWSTR>(wsFilename.c_str()), &mi, MMIO_READWRITE);
if (file == nullptr)
{
fprintf(stderr, "mmioOpen(\"%ls\", ...) failed. wErrorRet == %u\n", filename, mi.wErrorRet);
return;
}
// descend into the RIFF/WAVE chunk
ckRIFF = { 0 };
ckRIFF.ckid = MAKEFOURCC('W', 'A', 'V', 'E'); // this is right for mmioDescend
result = mmioDescend(file, &ckRIFF, nullptr, MMIO_FINDRIFF);
if (result != MMSYSERR_NOERROR)
{
fprintf(stderr, "mmioDescend(\"WAVE\") failed: MMSYSERR = %u\n", result);
return;
}
// descend into the fact chunk
MMCKINFO ckFact = { 0 };
ckFact.ckid = MAKEFOURCC('f', 'a', 'c', 't');
result = mmioDescend(file, &ckFact, &ckRIFF, MMIO_FINDCHUNK);
if (result != MMSYSERR_NOERROR) {
fprintf(stderr, "mmioDescend(\"fact\") failed: MMSYSERR = %u\n", result);
return;
}
// write the correct data to the fact chunk
LONG lBytesWritten = mmioWrite(file, reinterpret_cast<PCHAR>(&frames), sizeof(frames));
if (lBytesWritten != sizeof(frames))
{
fprintf(stderr, "Updating the fact chunk wrote %u bytes; expected %u\n", lBytesWritten, (UINT32)sizeof(frames));
return;
}
// ascend out of the fact chunk
result = mmioAscend(file, &ckFact, 0);
if (result != MMSYSERR_NOERROR)
fprintf(stderr, "mmioAscend(\"fact\") failed: MMSYSERR = %u\n", result);
}
//HRESULT LoopbackCapture(
// IMMDevice *pMMDevice,
// bool bInt16,
// HANDLE hStartedEvent,
// HANDLE hStopEvent,
// PUINT32 pnFrames,
// HMMIO hFile,
// AudioBuffer *pBuffer
//)
HRESULT LoopbackCapture::Process()
{
HRESULT hr;
// activate an IAudioClient
IAudioClient *pAudioClient;
hr = pMMDevice->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL, NULL,
(void**)&pAudioClient
);
if (FAILED(hr)) {
printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return hr;
}
// get the default device periodicity
REFERENCE_TIME hnsDefaultDevicePeriod;
hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, NULL);
if (FAILED(hr)) {
printf("IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
// get the default device format
WAVEFORMATEX *pwfx;
hr = pAudioClient->GetMixFormat(&pwfx);
if (FAILED(hr)) {
printf("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return hr;
}
if (pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
{
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
//pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
printf("WAVE_FORMAT_EXTENSIBLE\n");
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat))
{
printf("float\n");
}//
else if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_PCM, pEx->SubFormat))
{
printf("PCM\n");
}//KSDATAFORMAT_SUBTYPE_WAVEFORMATEX
else if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_WAVEFORMATEX, pEx->SubFormat))
{
printf("WAVEFORMATEX\n");
}
}
if (bInt16) {
// coerce int-16 wave format
// can do this in-place since we're not changing the size of the format
// also, the engine will auto-convert from float to int for us
switch (pwfx->wFormatTag) {
case WAVE_FORMAT_IEEE_FLOAT:
pwfx->wFormatTag = WAVE_FORMAT_PCM;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
pEx->Samples.wValidBitsPerSample = 16;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
} else {
printf("Don't know how to coerce mix format to int-16\n");
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
break;
default:
printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", pwfx->wFormatTag);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
MMCKINFO ckRIFF = {0};
MMCKINFO ckData = {0};
if (hFile!=NULL)
hr = WriteWaveHeader(hFile, pwfx, &ckRIFF, &ckData);
if (pBuffer)
{
bool isFloat = false;
switch (pwfx->wFormatTag) {
case WAVE_FORMAT_IEEE_FLOAT:
isFloat = true;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
isFloat = true;
}
}
break;
default:
break;
}
pBuffer->SetAudioInfo(pwfx->nSamplesPerSec,pwfx->nBlockAlign,pwfx->nChannels,pwfx->wBitsPerSample,isFloat);
}
if (FAILED(hr)) {
// WriteWaveHeader does its own logging
CoTaskMemFree(pwfx);
pAudioClient->Release();
return hr;
}
// create a periodic waitable timer
HANDLE hWakeUp = CreateWaitableTimer(NULL, FALSE, NULL);
if (NULL == hWakeUp) {
DWORD dwErr = GetLastError();
printf("CreateWaitableTimer failed: last error = %u\n", dwErr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
UINT32 nBlockAlign = pwfx->nBlockAlign;
UINT32 nChannels = pwfx->nChannels;
nFrames = 0;
// call IAudioClient::Initialize
// note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
// do not work together...
// the "data ready" event never gets set
// so we're going to do a timer-driven loop
hr = pAudioClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_LOOPBACK,
0, 0, pwfx, 0
);
if (FAILED(hr)) {
printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
CoTaskMemFree(pwfx);
// activate an IAudioCaptureClient
IAudioCaptureClient *pAudioCaptureClient;
hr = pAudioClient->GetService(
__uuidof(IAudioCaptureClient),
(void**)&pAudioCaptureClient
);
if (FAILED(hr)) {
printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
// register with MMCSS
DWORD nTaskIndex = 0;
HANDLE hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
if (NULL == hTask) {
DWORD dwErr = GetLastError();
printf("AvSetMmThreadCharacteristics failed: last error = %u\n", dwErr);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
// set the waitable timer
LARGE_INTEGER liFirstFire;
liFirstFire.QuadPart = -hnsDefaultDevicePeriod / 2; // negative means relative time
LONG lTimeBetweenFires = (LONG)hnsDefaultDevicePeriod / 2 / (10 * 1000); // convert to milliseconds
BOOL bOK = SetWaitableTimer(
hWakeUp,
&liFirstFire,
lTimeBetweenFires,
NULL, NULL, FALSE
);
if (!bOK) {
DWORD dwErr = GetLastError();
printf("SetWaitableTimer failed: last error = %u\n", dwErr);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
// call IAudioClient::Start
hr = pAudioClient->Start();
if (FAILED(hr)) {
printf("IAudioClient::Start failed: hr = 0x%08x\n", hr);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
SetEvent(hStartedEvent);
// loopback capture loop
HANDLE waitArray[2] = { hStopEvent, hWakeUp };
DWORD dwWaitResult;
DWORD immdState;
bool bDone = false;
bool bFirstPacket = true;
for (UINT32 nPasses = 0; !bDone; nPasses++) {
dwWaitResult = WaitForMultipleObjects(
ARRAYSIZE(waitArray), waitArray,
FALSE, INFINITE
);
if (WAIT_OBJECT_0 == dwWaitResult) {
//printf("Received stop event after %u passes and %u frames\n", nPasses, nFrames);
bDone = true;
continue; // exits loop
}
if (WAIT_OBJECT_0 + 1 != dwWaitResult) {
printf("Unexpected WaitForMultipleObjects return value %u on pass %u after %u frames\n", dwWaitResult, nPasses, nFrames);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
printf("'");
// got a "wake up" event - see if there's data
UINT32 nNextPacketSize;
hr = pAudioCaptureClient->GetNextPacketSize(&nNextPacketSize);
if (FAILED(hr)) {
if (hr == AUDCLNT_E_SERVICE_NOT_RUNNING)
printf("AUDCLNT_E_SERVICE_NOT_RUNNING : \n");
else if (hr == AUDCLNT_E_DEVICE_INVALIDATED)
printf("AUDCLNT_E_DEVICE_INVALIDATED : \n");
else
printf("UNKNOWN ERROR!!! : \n");
printf("IAudioCaptureClient::GetNextPacketSize failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, nFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
if (0 == nNextPacketSize) {
// no data yet
continue;
}
// get the captured data
BYTE *pData;
UINT32 nNumFramesToRead;
DWORD dwFlags;
hr = pAudioCaptureClient->GetBuffer(
&pData,
&nNumFramesToRead,
&dwFlags,
NULL,
NULL
);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetBuffer failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, nFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
if (bFirstPacket && AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY == dwFlags) {
printf("Probably spurious glitch reported on first packet\n");
}
else if (dwFlags & AUDCLNT_BUFFERFLAGS_SILENT)
{
printf("#");
}
else if (dwFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
{
printf("!");
}
else if (0 != dwFlags) {
printf("IAudioCaptureClient::GetBuffer set flags to 0x%08x on pass %u after %u frames\n", dwFlags, nPasses, nFrames);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
if (0 == nNumFramesToRead) {
// no data yet
continue;
}
//if (0 == nNumFramesToRead) {
// printf("IAudioCaptureClient::GetBuffer said to read 0 frames on pass %u after %u frames\n", nPasses, nFrames);
// pAudioClient->Stop();
// CancelWaitableTimer(hWakeUp);
// AvRevertMmThreadCharacteristics(hTask);
// pAudioCaptureClient->Release();
// CloseHandle(hWakeUp);
// pAudioClient->Release();
// return E_UNEXPECTED;
//}
LONG lBytesToWrite = nNumFramesToRead * nBlockAlign;
#pragma prefast(suppress: __WARNING_INCORRECT_ANNOTATION, "IAudioCaptureClient::GetBuffer SAL annotation implies a 1-byte buffer")
if (hFile!=NULL)
{
LONG lBytesWritten = mmioWrite(hFile, reinterpret_cast<PCHAR>(pData), lBytesToWrite);
if (lBytesToWrite != lBytesWritten) {
printf("mmioWrite wrote %u bytes on pass %u after %u frames: expected %u bytes\n", lBytesWritten, nPasses, nFrames, lBytesToWrite);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
if (pBuffer)
{
//switch (nBlockAlign/nChannels)
//{
//case 1:
// ShowPCM((unsigned char*)pData,nNumFramesToRead,nChannels,1024,60,"SYS_Byte");
// break;
//case 2:
// ShowPCM((short*)pData,nNumFramesToRead,nChannels,1024,60,"SYS_Short");
// break;
//case 4:
// ShowPCM((int*)pData,nNumFramesToRead,nChannels,1024,60,"SYS_Int");
// //ShowPCM((float*)pData,nNumFramesToRead,nChannels,1024,60,"SYS_float");
// break;
//}
pBuffer->PushBuffer(pData,lBytesToWrite);
}
nFrames += nNumFramesToRead;
hr = pAudioCaptureClient->ReleaseBuffer(nNumFramesToRead);
if (FAILED(hr)) {
printf("IAudioCaptureClient::ReleaseBuffer failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, nFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
bFirstPacket = false;
} // capture loop
if (hFile!=NULL)
hr = FinishWaveFile(hFile, &ckData, &ckRIFF);
if (FAILED(hr)) {
// FinishWaveFile does it's own logging
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
DWORD CWASAPICapture::DoCaptureThread()
{
HANDLE mmcssHandle = NULL;
IXAudio2* xaudio = 0;
IXAudio2MasteringVoice* mastering_voice = 0;
IXAudio2SourceVoice* source_voice = 0;
try {
bool stillPlaying = true;
DWORD mmcssTaskIndex = 0;
HRESULT hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
if (FAILED(hr))
{
printf_s("Unable to initialize COM in render thread: %x\n", hr);
return hr;
}
mmcssHandle = AvSetMmThreadCharacteristics(L"Audio", &mmcssTaskIndex);
if (mmcssHandle == NULL)
{
printf_s("Unable to enable MMCSS on capture thread: %d\n", GetLastError());
}
//
// XAudioの初期化
//
{
UINT32 flags = 0;
#ifdef _DEBUG
flags |= XAUDIO2_DEBUG_ENGINE;
#endif
if( FAILED( hr = XAudio2Create( &xaudio, flags ) ) )
throw "XAudio2Create";
// Create a mastering voice
if( FAILED( hr = xaudio->CreateMasteringVoice( &mastering_voice ) ) )
throw "CreateMasteringVoice";
// WAVファイルのWAVEFORMATEXを使ってSourceVoiceを作成
if( FAILED( xaudio->CreateSourceVoice( &source_voice, MixFormat() ) ) )
throw "CreateSourceVoice";
// 再生
source_voice->Start();
}
while (stillPlaying)
{
HRESULT hr;
//
// In Timer Driven mode, we want to wait for half the desired latency in milliseconds.
//
// That way we'll wake up half way through the processing period to pull the
// next set of samples from the engine.
//
DWORD waitResult = WaitForSingleObject(_ShutdownEvent, _EngineLatencyInMS / 2);
switch (waitResult)
{
case WAIT_OBJECT_0 + 0: // _ShutdownEvent
stillPlaying = false; // We're done, exit the loop.
break;
case WAIT_TIMEOUT: // Timeout
//
// We need to retrieve the next buffer of samples from the audio capturer.
//
BYTE *pData;
UINT32 framesAvailable;
DWORD flags;
//
// Find out how much capture data is available. We need to make sure we don't run over the length
// of our capture buffer. We'll discard any samples that don't fit in the buffer.
//
hr = _CaptureClient->GetBuffer(&pData, &framesAvailable, &flags, NULL, NULL);
if (SUCCEEDED(hr))
{
UINT32 framesToCopy = min(framesAvailable, static_cast<UINT32>((_CaptureBufferSize - _CurrentCaptureIndex) / _FrameSize));
if (framesToCopy != 0)
{
//
// The flags on capture tell us information about the data.
//
// We only really care about the silent flag since we want to put frames of silence into the buffer
// when we receive silence. We rely on the fact that a logical bit 0 is silence for both float and int formats.
//
if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
{
//
// Fill 0s from the capture buffer to the output buffer.
//
ZeroMemory(&_CaptureBuffer[_CurrentCaptureIndex], framesToCopy*_FrameSize);
}
else
{
//
// Copy data from the audio engine buffer to the output buffer.
//
CopyMemory(&_CaptureBuffer[_CurrentCaptureIndex], pData, framesToCopy*_FrameSize);
// SourceVoiceにデータを送信
XAUDIO2_BUFFER buffer = { 0 };
buffer.AudioBytes = framesToCopy * _FrameSize; //バッファのバイト数
buffer.pAudioData = &pData[ 0 ]; //バッファの先頭アドレス
source_voice->SubmitSourceBuffer( &buffer );
}
//
// Bump the capture buffer pointer.
//
_CurrentCaptureIndex += framesToCopy*_FrameSize;
}
hr = _CaptureClient->ReleaseBuffer(framesAvailable);
if (FAILED(hr))
{
printf_s("Unable to release capture buffer: %x!\n", hr);
}
}
break;
}
}
}
catch( const char* e )
{
std::cout << e << std::endl;
}
// Cleanup XAudio2
if( mastering_voice != 0 ) {
// ここで落ちる
//mastering_voice->DestroyVoice();
mastering_voice = 0;
}
if( xaudio != 0 ) {
// ここでも落ちる
//xaudio->Release();
xaudio = 0;
}
AvRevertMmThreadCharacteristics(mmcssHandle);
CoUninitialize();
return 0;
}
HRESULT LoopbackCapture(
IMMDevice *pMMDevice,
bool bInt16,
HANDLE hStartedEvent,
HANDLE hStopEvent,
PUINT32 pnFrames,
bool bMono,
INT32 iSampleRateDivisor
) {
HRESULT hr;
SimpleTcpServer server;
// Wait for client connection before attempting any audio capture
server.setup();
server.waitForClient();
// activate an IAudioClient
IAudioClient *pAudioClient;
hr = pMMDevice->Activate(
__uuidof(IAudioClient),
CLSCTX_ALL, NULL,
(void**)&pAudioClient
);
if (FAILED(hr)) {
printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return hr;
}
// get the default device periodicity
REFERENCE_TIME hnsDefaultDevicePeriod;
hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, NULL);
if (FAILED(hr)) {
printf("IAudioClient::GetDevicePeriod failed: hr = 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
// get the default device format
WAVEFORMATEX *pwfx;
hr = pAudioClient->GetMixFormat(&pwfx);
if (FAILED(hr)) {
printf("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return hr;
}
if (bInt16) {
// coerce int-16 wave format
// can do this in-place since we're not changing the size of the format
// also, the engine will auto-convert from float to int for us
switch (pwfx->wFormatTag) {
case WAVE_FORMAT_IEEE_FLOAT:
pwfx->wFormatTag = WAVE_FORMAT_PCM;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
pEx->Samples.wValidBitsPerSample = 16;
pwfx->wBitsPerSample = 16;
pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
} else {
printf("Don't know how to coerce mix format to int-16\n");
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
break;
default:
printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", pwfx->wFormatTag);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return E_UNEXPECTED;
}
}
// create a periodic waitable timer
HANDLE hWakeUp = CreateWaitableTimer(NULL, FALSE, NULL);
if (NULL == hWakeUp) {
DWORD dwErr = GetLastError();
printf("CreateWaitableTimer failed: last error = %u\n", dwErr);
CoTaskMemFree(pwfx);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
UINT32 nBlockAlign = pwfx->nBlockAlign;
UINT32 nBufferSize;
*pnFrames = 0;
// call IAudioClient::Initialize
// note that AUDCLNT_STREAMFLAGS_LOOPBACK and AUDCLNT_STREAMFLAGS_EVENTCALLBACK
// do not work together...
// the "data ready" event never gets set
// so we're going to do a timer-driven loop
hr = pAudioClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_LOOPBACK,
0, 0, pwfx, 0
);
if (FAILED(hr)) {
printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
CoTaskMemFree(pwfx);
// Get the buffer size
hr = pAudioClient->GetBufferSize(&nBufferSize);
if (FAILED(hr)) {
printf("IAudioClient::GetBufferSize failed: hr = 0x%08x\n", hr);
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
// Configure the server. The buffer size returned is in frames
// so assume stereo, 16 bits per sample to convert from frames to bytes
server.configure(
bMono,
iSampleRateDivisor,
nBufferSize * 2 * 2);
// activate an IAudioCaptureClient
IAudioCaptureClient *pAudioCaptureClient;
hr = pAudioClient->GetService(
__uuidof(IAudioCaptureClient),
(void**)&pAudioCaptureClient
);
if (FAILED(hr)) {
printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
// register with MMCSS
DWORD nTaskIndex = 0;
HANDLE hTask = AvSetMmThreadCharacteristics(L"Capture", &nTaskIndex);
if (NULL == hTask) {
DWORD dwErr = GetLastError();
printf("AvSetMmThreadCharacteristics failed: last error = %u\n", dwErr);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
// set the waitable timer
LARGE_INTEGER liFirstFire;
liFirstFire.QuadPart = -hnsDefaultDevicePeriod / 2; // negative means relative time
LONG lTimeBetweenFires = (LONG)hnsDefaultDevicePeriod / 2 / (10 * 1000); // convert to milliseconds
BOOL bOK = SetWaitableTimer(
hWakeUp,
&liFirstFire,
lTimeBetweenFires,
NULL, NULL, FALSE
);
if (!bOK) {
DWORD dwErr = GetLastError();
printf("SetWaitableTimer failed: last error = %u\n", dwErr);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return HRESULT_FROM_WIN32(dwErr);
}
// call IAudioClient::Start
hr = pAudioClient->Start();
if (FAILED(hr)) {
printf("IAudioClient::Start failed: hr = 0x%08x\n", hr);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
SetEvent(hStartedEvent);
// loopback capture loop
HANDLE waitArray[2] = { hStopEvent, hWakeUp };
DWORD dwWaitResult;
bool bDone = false;
for (UINT32 nPasses = 0; !bDone; nPasses++) {
// drain data while it is available
UINT32 nNextPacketSize;
for (
hr = pAudioCaptureClient->GetNextPacketSize(&nNextPacketSize);
SUCCEEDED(hr) && nNextPacketSize > 0;
hr = pAudioCaptureClient->GetNextPacketSize(&nNextPacketSize)
) {
// get the captured data
BYTE *pData;
UINT32 nNumFramesToRead;
DWORD dwFlags;
hr = pAudioCaptureClient->GetBuffer(
&pData,
&nNumFramesToRead,
&dwFlags,
NULL,
NULL
);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetBuffer failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, *pnFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
#ifdef _DEBUG
if (0 != dwFlags) {
printf("[ignoring] IAudioCaptureClient::GetBuffer set flags to 0x%08x on pass %u after %u frames\n", dwFlags, nPasses, *pnFrames);
}
#endif
if (0 == nNumFramesToRead) {
printf("IAudioCaptureClient::GetBuffer said to read 0 frames on pass %u after %u frames\n", nPasses, *pnFrames);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
LONG lBytesToWrite = nNumFramesToRead * nBlockAlign;
if (server.sendData(reinterpret_cast<const char*>(pData), lBytesToWrite) != 0) {
printf("Error sending data to peer\n");
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
*pnFrames += nNumFramesToRead;
hr = pAudioCaptureClient->ReleaseBuffer(nNumFramesToRead);
if (FAILED(hr)) {
printf("IAudioCaptureClient::ReleaseBuffer failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, *pnFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
}
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetNextPacketSize failed on pass %u after %u frames: hr = 0x%08x\n", nPasses, *pnFrames, hr);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return hr;
}
dwWaitResult = WaitForMultipleObjects(
ARRAYSIZE(waitArray), waitArray,
FALSE, INFINITE
);
if (WAIT_OBJECT_0 == dwWaitResult) {
printf("Received stop event after %u passes and %u frames\n", nPasses, *pnFrames);
bDone = true;
continue; // exits loop
}
if (WAIT_OBJECT_0 + 1 != dwWaitResult) {
printf("Unexpected WaitForMultipleObjects return value %u on pass %u after %u frames\n", dwWaitResult, nPasses, *pnFrames);
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
return E_UNEXPECTED;
}
} // capture loop
pAudioClient->Stop();
CancelWaitableTimer(hWakeUp);
AvRevertMmThreadCharacteristics(hTask);
pAudioCaptureClient->Release();
CloseHandle(hWakeUp);
pAudioClient->Release();
server.shutdown();
return hr;
}
