BlankAudioPlayback(CTSTR lpDevice)
{
const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
const IID IID_IAudioClient = __uuidof(IAudioClient);
const IID IID_IAudioRenderClient = __uuidof(IAudioRenderClient);
HRESULT err;
err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
if(FAILED(err))
CrashError(TEXT("Could not create IMMDeviceEnumerator: 0x%08lx"), err);
if (scmpi(lpDevice, TEXT("Default")) == 0)
err = mmEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &mmDevice);
else
err = mmEnumerator->GetDevice(lpDevice, &mmDevice);
if(FAILED(err))
CrashError(TEXT("Could not create IMMDevice"));
err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
if(FAILED(err))
CrashError(TEXT("Could not create IAudioClient"));
WAVEFORMATEX *pwfx;
err = mmClient->GetMixFormat(&pwfx);
if(FAILED(err))
CrashError(TEXT("Could not get mix format from audio client"));
UINT inputBlockSize = pwfx->nBlockAlign;
err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, 0, ConvertMSTo100NanoSec(1000), 0, pwfx, NULL);
if(FAILED(err))
CrashError(TEXT("Could not initialize audio client, error = %08lX"), err);
err = mmClient->GetService(IID_IAudioRenderClient, (void**)&mmRender);
if(FAILED(err))
CrashError(TEXT("Could not get audio render client"));
//----------------------------------------------------------------
UINT bufferFrameCount;
err = mmClient->GetBufferSize(&bufferFrameCount);
if(FAILED(err))
CrashError(TEXT("Could not get audio buffer size"));
BYTE *lpData;
err = mmRender->GetBuffer(bufferFrameCount, &lpData);
if(FAILED(err))
CrashError(TEXT("Could not get audio buffer"));
zero(lpData, bufferFrameCount*inputBlockSize);
mmRender->ReleaseBuffer(bufferFrameCount, 0);//AUDCLNT_BUFFERFLAGS_SILENT); //probably better if it doesn't know
if(FAILED(mmClient->Start()))
CrashError(TEXT("Could not start audio source"));
}
//
// Initialize WASAPI in event driven mode, associate the audio client with our samples ready event handle, retrieve
// a capture client for the transport, create the capture thread and start the audio engine.
//
bool CWASAPICapture::InitializeAudioEngine()
{
HRESULT hr = _AudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_NOPERSIST, _EngineLatencyInMS*10000, 0, MixFormat(), NULL);
PersistentAssert(SUCCEEDED(hr), "_AudioClient->Initialize failed");
//
// Retrieve the buffer size for the audio client.
//
hr = _AudioClient->GetBufferSize(&_BufferSize);
PersistentAssert(SUCCEEDED(hr), "_AudioClient->GetBufferSize failed");
hr = _AudioClient->GetService(IID_PPV_ARGS(&_CaptureClient));
PersistentAssert(SUCCEEDED(hr), "_AudioClient->GetService failed");
return true;
}
HRESULT __stdcall getservice_patch(IAudioClient* this_, REFIID riid, void** ppv)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetService(riid, ppv);
((DWORD_PTR**)this_)[0] = old_vftptr;
// renderclient list has 1:1 mapping to audioclient
if(hr == S_OK)
{
if(riid == __uuidof(IAudioRenderClient))
{
IAudioRenderClient* host = *((IAudioRenderClient**)ppv);
patch_iaudiorenderclient(host, *((WORD***)this_)[0][18]);
for(iaudioclient_duplicate* next = get_duplicate(this_)->next;
next != NULL; next = next->next)
{
IAudioRenderClient* renderclient = NULL;
next->proxy->GetService(riid, (void**)&renderclient);
get_duplicate(host)->add(renderclient);
}
}
else if(riid == __uuidof(IAudioStreamVolume))
{
IAudioStreamVolume* host = *((IAudioStreamVolume**)ppv);
patch_iaudiostreamvolume(host);
for(iaudioclient_duplicate* next = get_duplicate(this_)->next;
next != NULL; next = next->next)
{
IAudioStreamVolume* streamvolume = NULL;
next->proxy->GetService(riid, (void**)&streamvolume);
if(streamvolume != NULL)
get_duplicate(host)->add(streamvolume);
}
}
}
return hr;
}
bool MMDeviceAudioSource::Initialize(bool bMic, CTSTR lpID)
{
const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
const IID IID_IAudioClient = __uuidof(IAudioClient);
const IID IID_IAudioCaptureClient = __uuidof(IAudioCaptureClient);
HRESULT err;
err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDeviceEnumerator = %08lX"), (BOOL)bMic, err);
return false;
}
bIsMic = bMic;
if (bIsMic) {
BOOL bMicSyncFixHack = GlobalConfig->GetInt(TEXT("Audio"), TEXT("UseMicSyncFixHack"));
angerThreshold = bMicSyncFixHack ? 40 : 1000;
}
if (scmpi(lpID, TEXT("Default")) == 0)
err = mmEnumerator->GetDefaultAudioEndpoint(bMic ? eCapture : eRender, bMic ? eCommunications : eConsole, &mmDevice);
else
err = mmEnumerator->GetDevice(lpID, &mmDevice);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDevice = %08lX"), (BOOL)bMic, err);
return false;
}
err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IAudioClient = %08lX"), (BOOL)bMic, err);
return false;
}
//-----------------------------------------------------------------
// get name
IPropertyStore *store;
if(SUCCEEDED(mmDevice->OpenPropertyStore(STGM_READ, &store)))
{
PROPVARIANT varName;
PropVariantInit(&varName);
if(SUCCEEDED(store->GetValue(PKEY_Device_FriendlyName, &varName)))
{
CWSTR wstrName = varName.pwszVal;
strDeviceName = wstrName;
}
store->Release();
}
if(bMic)
{
Log(TEXT("------------------------------------------"));
Log(TEXT("Using auxilary audio input: %s"), GetDeviceName());
bUseQPC = GlobalConfig->GetInt(TEXT("Audio"), TEXT("UseMicQPC")) != 0;
if (bUseQPC)
Log(TEXT("Using Mic QPC timestamps"));
}
else
{
Log(TEXT("------------------------------------------"));
Log(TEXT("Using desktop audio input: %s"), GetDeviceName());
bUseVideoTime = AppConfig->GetInt(TEXT("Audio"), TEXT("SyncToVideoTime")) != 0;
SetTimeOffset(GlobalConfig->GetInt(TEXT("Audio"), TEXT("GlobalAudioTimeAdjust")));
}
//-----------------------------------------------------------------
// get format
WAVEFORMATEX *pwfx;
err = mmClient->GetMixFormat(&pwfx);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get mix format from audio client = %08lX"), (BOOL)bMic, err);
return false;
}
bool bFloat;
UINT inputChannels;
UINT inputSamplesPerSec;
UINT inputBitsPerSample;
UINT inputBlockSize;
DWORD inputChannelMask = 0;
WAVEFORMATEXTENSIBLE *wfext = NULL;
//the internal audio engine should always use floats (or so I read), but I suppose just to be safe better check
if(pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
{
wfext = (WAVEFORMATEXTENSIBLE*)pwfx;
inputChannelMask = wfext->dwChannelMask;
if(wfext->SubFormat != KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
return false;
}
}
else if(pwfx->wFormatTag != WAVE_FORMAT_IEEE_FLOAT)
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
return false;
}
bFloat = true;
inputChannels = pwfx->nChannels;
inputBitsPerSample = 32;
inputBlockSize = pwfx->nBlockAlign;
inputSamplesPerSec = pwfx->nSamplesPerSec;
sampleWindowSize = (inputSamplesPerSec/100);
DWORD flags = bMic ? 0 : AUDCLNT_STREAMFLAGS_LOOPBACK;
err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, ConvertMSTo100NanoSec(5000), 0, pwfx, NULL);
//err = AUDCLNT_E_UNSUPPORTED_FORMAT;
if (err == AUDCLNT_E_UNSUPPORTED_FORMAT) { //workaround for razer kraken headset (bad drivers)
pwfx->nBlockAlign = 2*pwfx->nChannels;
pwfx->nAvgBytesPerSec = inputSamplesPerSec*pwfx->nBlockAlign;
pwfx->wBitsPerSample = 16;
wfext->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wfext->Samples.wValidBitsPerSample = 16;
bConvert = true;
err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, ConvertMSTo100NanoSec(5000), 0, pwfx, NULL);
}
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not initialize audio client, result = %08lX"), (BOOL)bMic, err);
return false;
}
//-----------------------------------------------------------------
// acquire services
err = mmClient->GetService(IID_IAudioCaptureClient, (void**)&mmCapture);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture client, result = %08lX"), (BOOL)bMic, err);
return false;
}
err = mmClient->GetService(__uuidof(IAudioClock), (void**)&mmClock);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture clock, result = %08lX"), (BOOL)bMic, err);
return false;
}
CoTaskMemFree(pwfx);
//-----------------------------------------------------------------
InitAudioData(bFloat, inputChannels, inputSamplesPerSec, inputBitsPerSample, inputBlockSize, inputChannelMask);
return true;
}
void PlayAudio()
{
REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC; // microseconds, so this is 1 seconds
REFERENCE_TIME hnsActualDuration;
HRESULT hr;
IMMDeviceEnumerator *pEnumerator = NULL;
IMMDevice *pDevice = NULL;
IAudioClient *pAudioClient = NULL;
IAudioRenderClient *pRenderClient = NULL;
WAVEFORMATEX *pwfx = NULL;
UINT32 bufferFrameCount;
UINT32 numFramesAvailable;
UINT32 numFramesPadding;
BYTE *pData;
DWORD flags = 0;
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(&pwfx);
EXIT_ON_ERROR(hr);
hr = pAudioClient->Initialize(
AUDCLNT_SHAREMODE_SHARED,
0,
hnsRequestedDuration,
0,
pwfx,
NULL);
EXIT_ON_ERROR(hr);
// Get the actual size of the allocated buffer.
hr = pAudioClient->GetBufferSize(&bufferFrameCount);
EXIT_ON_ERROR(hr);
hr = pAudioClient->GetService(
IID_IAudioRenderClient,
(void**)&pRenderClient);
EXIT_ON_ERROR(hr);
// Grab the entire buffer for the initial fill operation.
hr = pRenderClient->GetBuffer(bufferFrameCount, &pData);
EXIT_ON_ERROR(hr);
// load initial data
hr = LoadAudioBuffer(bufferFrameCount, pData, pwfx, &flags);
EXIT_ON_ERROR(hr);
hr = pRenderClient->ReleaseBuffer(bufferFrameCount, flags);
EXIT_ON_ERROR(hr);
// Calculate the actual duration of the allocated buffer.
hnsActualDuration = (REFERENCE_TIME)((double)REFTIMES_PER_SEC * bufferFrameCount / pwfx->nSamplesPerSec);
hr = pAudioClient->Start(); // Start playing.
EXIT_ON_ERROR(hr);
// Each loop fills about half of the shared buffer.
while (flags != AUDCLNT_BUFFERFLAGS_SILENT)
{
// Sleep for half the buffer duration.
Sleep((DWORD)(hnsActualDuration/REFTIMES_PER_MILLISEC/2));
// See how much buffer space is available.
hr = pAudioClient->GetCurrentPadding(&numFramesPadding);
EXIT_ON_ERROR(hr)
numFramesAvailable = bufferFrameCount - numFramesPadding;
// Grab all the available space in the shared buffer.
hr = pRenderClient->GetBuffer(numFramesAvailable, &pData);
EXIT_ON_ERROR(hr)
// Get next 1/2-second of data from the audio source.
hr = LoadAudioBuffer(numFramesAvailable, pData, pwfx, &flags);
EXIT_ON_ERROR(hr)
hr = pRenderClient->ReleaseBuffer(numFramesAvailable, flags);
EXIT_ON_ERROR(hr)
}
// Wait for last data in buffer to play before stopping.
Sleep((DWORD)(hnsActualDuration/REFTIMES_PER_MILLISEC/2));
hr = pAudioClient->Stop(); // Stop playing.
EXIT_ON_ERROR(hr);
Exit:
CoTaskMemFree(pwfx);
SAFE_RELEASE(pEnumerator);
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pAudioClient);
SAFE_RELEASE(pRenderClient);
}
bool MMDeviceAudioSource::Initialize(bool bMic, CTSTR lpID)
{
const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
const IID IID_IAudioClient = __uuidof(IAudioClient);
const IID IID_IAudioCaptureClient = __uuidof(IAudioCaptureClient);
HRESULT err;
err = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&mmEnumerator);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDeviceEnumerator = %08lX"), (BOOL)bMic, err);
return false;
}
if(bMic)
err = mmEnumerator->GetDevice(lpID, &mmDevice);
else
err = mmEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &mmDevice);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IMMDevice = %08lX"), (BOOL)bMic, err);
return false;
}
err = mmDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&mmClient);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not create IAudioClient = %08lX"), (BOOL)bMic, err);
return false;
}
WAVEFORMATEX *pwfx;
err = mmClient->GetMixFormat(&pwfx);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get mix format from audio client = %08lX"), (BOOL)bMic, err);
return false;
}
String strName = GetDeviceName();
if(bMic)
{
Log(TEXT("------------------------------------------"));
Log(TEXT("Using auxilary audio input: %s"), strName.Array());
}
//the internal audio engine should always use floats (or so I read), but I suppose just to be safe better check
if(pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE)
{
WAVEFORMATEXTENSIBLE *wfext = (WAVEFORMATEXTENSIBLE*)pwfx;
inputChannelMask = wfext->dwChannelMask;
if(wfext->SubFormat != KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
return false;
}
}
else if(pwfx->wFormatTag != WAVE_FORMAT_IEEE_FLOAT)
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Unsupported wave format"), (BOOL)bMic);
return false;
}
inputChannels = pwfx->nChannels;
inputBitsPerSample = 32;
inputBlockSize = pwfx->nBlockAlign;
inputSamplesPerSec = pwfx->nSamplesPerSec;
DWORD flags = bMic ? 0 : AUDCLNT_STREAMFLAGS_LOOPBACK;
err = mmClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flags, ConvertMSTo100NanoSec(5000), 0, pwfx, NULL);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not initialize audio client, result = %08lX"), (BOOL)bMic, err);
return false;
}
err = mmClient->GetService(IID_IAudioCaptureClient, (void**)&mmCapture);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture client, result = %08lX"), (BOOL)bMic, err);
return false;
}
err = mmClient->GetService(__uuidof(IAudioClock), (void**)&mmClock);
if(FAILED(err))
{
AppWarning(TEXT("MMDeviceAudioSource::Initialize(%d): Could not get audio capture clock, result = %08lX"), (BOOL)bMic, err);
return false;
}
CoTaskMemFree(pwfx);
//-------------------------------------------------------------------------
if(inputSamplesPerSec != 44100)
{
int errVal;
int converterType = AppConfig->GetInt(TEXT("Audio"), TEXT("UseHighQualityResampling"), FALSE) ? SRC_SINC_FASTEST : SRC_LINEAR;
resampler = src_new(converterType, 2, &errVal);//SRC_SINC_FASTEST//SRC_ZERO_ORDER_HOLD
if(!resampler)
{
CrashError(TEXT("MMDeviceAudioSource::Initialize(%d): Could not initiate resampler"), (BOOL)bMic);
return false;
}
resampleRatio = 44100.0 / double(inputSamplesPerSec);
bResample = true;
//----------------------------------------------------
// hack to get rid of that weird first quirky resampled packet size
// (always returns a non-441 sized packet on the first resample)
SRC_DATA data;
data.src_ratio = resampleRatio;
List<float> blankBuffer;
blankBuffer.SetSize(inputSamplesPerSec/100*2);
data.data_in = blankBuffer.Array();
data.input_frames = inputSamplesPerSec/100;
UINT frameAdjust = UINT((double(data.input_frames) * resampleRatio) + 1.0);
UINT newFrameSize = frameAdjust*2;
tempResampleBuffer.SetSize(newFrameSize);
data.data_out = tempResampleBuffer.Array();
data.output_frames = frameAdjust;
data.end_of_input = 0;
int err = src_process(resampler, &data);
nop();
}
//-------------------------------------------------------------------------
if(inputChannels > 2)
{
if(inputChannelMask == 0)
{
switch(inputChannels)
{
case 3: inputChannelMask = KSAUDIO_SPEAKER_2POINT1; break;
case 4: inputChannelMask = KSAUDIO_SPEAKER_QUAD; break;
case 5: inputChannelMask = KSAUDIO_SPEAKER_4POINT1; break;
case 6: inputChannelMask = KSAUDIO_SPEAKER_5POINT1; break;
case 8: inputChannelMask = KSAUDIO_SPEAKER_7POINT1; break;
}
}
switch(inputChannelMask)
{
case KSAUDIO_SPEAKER_QUAD: Log(TEXT("Using quad speaker setup")); break; //ocd anyone?
case KSAUDIO_SPEAKER_2POINT1: Log(TEXT("Using 2.1 speaker setup")); break;
case KSAUDIO_SPEAKER_4POINT1: Log(TEXT("Using 4.1 speaker setup")); break;
case KSAUDIO_SPEAKER_SURROUND: Log(TEXT("Using basic surround speaker setup")); break;
case KSAUDIO_SPEAKER_5POINT1: Log(TEXT("Using 5.1 speaker setup")); break;
case KSAUDIO_SPEAKER_5POINT1_SURROUND: Log(TEXT("Using 5.1 surround speaker setup")); break;
case KSAUDIO_SPEAKER_7POINT1: Log(TEXT("Using 7.1 speaker setup (experimental)")); break;
case KSAUDIO_SPEAKER_7POINT1_SURROUND: Log(TEXT("Using 7.1 surround speaker setup (experimental)")); break;
default:
Log(TEXT("Using unknown speaker setup: 0x%lX"), inputChannelMask);
CrashError(TEXT("Speaker setup not yet implemented -- dear god of all the audio APIs, the one I -have- to use doesn't support resampling or downmixing. fabulous."));
break;
}
}
return true;
}
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;
}
int main(int argc, char *argv[])
{
CoInitialize(nullptr);
listDevices();
IAudioClient *pAudioClient;
IMMDevice *device;
getDefaultDevice(&device);
HRESULT hr = device->Activate(__uuidof(IAudioClient),
CLSCTX_ALL, nullptr, (void**)&pAudioClient);
if (FAILED(hr)) {
printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return hr;
}
REFERENCE_TIME hnsDefaultDevicePeriod;
hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, nullptr);
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;
}
DVAR(pwfx->wFormatTag);
DVAR(pwfx->wBitsPerSample);
DVAR(pwfx->nBlockAlign);
DVAR(pwfx->nAvgBytesPerSec);
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;
}
DVAR(pwfx->wFormatTag);
DVAR(pwfx->wBitsPerSample);
DVAR(pwfx->nBlockAlign);
DVAR(pwfx->nAvgBytesPerSec);
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);
pAudioClient->Release();
return hr;
}
IAudioCaptureClient *pAudioCaptureClient;
hr = pAudioClient->GetService(__uuidof(IAudioCaptureClient), (void**)&pAudioCaptureClient);
if (FAILED(hr)) {
printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
pAudioClient->Release();
return hr;
}
hr = pAudioClient->Start();
if (FAILED(hr)) {
printf("IAudioClient::Start failed: hr = 0x%08x\n", hr);
pAudioCaptureClient->Release();
pAudioClient->Release();
return hr;
}
for (int i = 0; i < 10; ++i)
{
UINT32 nNextPacketSize;
hr = pAudioCaptureClient->GetNextPacketSize(&nNextPacketSize);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetNextPacketSize failed on pass %u after %u frames: hr = 0x%08x\n", 0, 0, hr);
pAudioClient->Stop();
pAudioCaptureClient->Release();
pAudioClient->Release();
return hr;
}
// get the captured data
BYTE *pData;
UINT32 nNumFramesToRead;
DWORD dwFlags;
hr = pAudioCaptureClient->GetBuffer(&pData, &nNumFramesToRead, &dwFlags, nullptr,
nullptr);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetBuffer failed on pass %u after %u frames: hr = 0x%08x\n", 0, 0, hr);
pAudioClient->Stop();
pAudioCaptureClient->Release();
pAudioClient->Release();
return hr;
}
DVAR(nNumFramesToRead);
// if (bFirstPacket && AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY == dwFlags) {
// printf("Probably spurious glitch reported on first packet\n");
if (0 != dwFlags && AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY != dwFlags) {
printf("IAudioCaptureClient::GetBuffer set flags to 0x%08x on pass %u after %u frames\n", dwFlags, 0, 0);
// pAudioClient->Stop();
// pAudioCaptureClient->Release();
// pAudioClient->Release();
// return E_UNEXPECTED;
}
else
DVAR((int)*pData);
if (0 == nNumFramesToRead) {
printf("IAudioCaptureClient::GetBuffer said to read 0 frames on pass %u after %u frames\n", 0, 0);
pAudioClient->Stop();
pAudioCaptureClient->Release();
pAudioClient->Release();
return E_UNEXPECTED;
}
UINT32 nBlockAlign = pwfx->nBlockAlign;
LONG lBytesToWrite = nNumFramesToRead * nBlockAlign;
hr = pAudioCaptureClient->ReleaseBuffer(nNumFramesToRead);
}
pAudioClient->Stop();
pAudioCaptureClient->Release();
pAudioClient->Release();
CoUninitialize();
return 0;
}
int _tmain(int argc, _TCHAR* argv[]) {
IMMDeviceEnumerator *enumerator = 0;
IMMDevice *device = 0;
FILE *f;
f=fopen("c:/1.wav","w");
CoInitialize(0);
CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator),
(void**) &enumerator);
enumerator->GetDefaultAudioEndpoint(eRender, eConsole, &device);
HANDLE processOutWrite, processOutRead, processInWrite, processInRead;
/*wchar_t processCommand[2000];
{
FILE* commandFile;
fopen_s(&commandFile, "command.txt", "r");
char cmd[2000];
fread(cmd, sizeof(char), 2000, commandFile);
fclose(commandFile);
size_t count;
mbstowcs_s(&count, processCommand, cmd, 2000);
}*/
/*{
//create pipes for plink process
SECURITY_ATTRIBUTES pipeAttributes = {0};
pipeAttributes.nLength = sizeof(SECURITY_ATTRIBUTES);
pipeAttributes.bInheritHandle = TRUE;
pipeAttributes.lpSecurityDescriptor= NULL;
CreatePipe(&processOutRead, &processOutWrite, &pipeAttributes, 0);
CreatePipe(&processInRead, &processInWrite, &pipeAttributes, 0);
STARTUPINFO startupInfo;
ZeroMemory(&startupInfo, sizeof(STARTUPINFO));
startupInfo.cb = sizeof(STARTUPINFO);
startupInfo.hStdError = processOutWrite;
startupInfo.hStdOutput = processOutWrite;
startupInfo.hStdInput = processInRead;
startupInfo.dwFlags |= STARTF_USESTDHANDLES;
PROCESS_INFORMATION processInfo = {0};
//launch process
CreateProcess(NULL, processCommand, NULL, NULL, TRUE, 0, NULL, NULL, &startupInfo, &processInfo);
//wait for plink to connect to minimze sound delay (magic number)
Sleep(2500);
}*/
HRESULT hr;
// activate an IAudioClient
IAudioClient *audioClient;
hr = device->Activate(__uuidof(IAudioClient), CLSCTX_ALL, NULL, (void**) &audioClient);
if (FAILED(hr)) {
printf("IMMDevice::Activate(IAudioClient) failed: hr = 0x%08x", hr);
return hr;
}
// get the default device format
WAVEFORMATEX *waveFormat;
hr = audioClient->GetMixFormat(&waveFormat);
if (FAILED(hr)) {
printf("IAudioClient::GetMixFormat failed: hr = 0x%08x\n", hr);
CoTaskMemFree(waveFormat);
audioClient->Release();
return hr;
}
// 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 (waveFormat->wFormatTag) {
case WAVE_FORMAT_IEEE_FLOAT:
waveFormat->wFormatTag = WAVE_FORMAT_PCM;
waveFormat->wBitsPerSample = 16;
waveFormat->nBlockAlign = waveFormat->nChannels * waveFormat->wBitsPerSample / 8;
waveFormat->nAvgBytesPerSec = waveFormat->nBlockAlign * waveFormat->nSamplesPerSec;
break;
case WAVE_FORMAT_EXTENSIBLE:
{
// naked scope for case-local variable
PWAVEFORMATEXTENSIBLE waveFormatEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(waveFormat);
if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, waveFormatEx->SubFormat)) {
waveFormatEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
waveFormatEx->Samples.wValidBitsPerSample = 16;
waveFormat->wBitsPerSample = 16;
waveFormat->nBlockAlign = waveFormat->nChannels * waveFormat->wBitsPerSample / 8;
waveFormat->nAvgBytesPerSec = waveFormat->nBlockAlign * waveFormat->nSamplesPerSec;
} else {
printf("Don't know how to coerce mix format to int-16\n");
CoTaskMemFree(waveFormat);
audioClient->Release();
return E_UNEXPECTED;
}
}
break;
default:
printf("Don't know how to coerce WAVEFORMATEX with wFormatTag = 0x%08x to int-16\n", waveFormat->wFormatTag);
CoTaskMemFree(waveFormat);
audioClient->Release();
return E_UNEXPECTED;
}
UINT32 blockAlign = waveFormat->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,
10000000, 0, waveFormat, 0
);
if (FAILED(hr)) {
printf("IAudioClient::Initialize failed: hr = 0x%08x\n", hr);
audioClient->Release();
return hr;
}
CoTaskMemFree(waveFormat);
// activate an IAudioCaptureClient
IAudioCaptureClient *audioCaptureClient;
hr = audioClient->GetService(__uuidof(IAudioCaptureClient), (void**) &audioCaptureClient);
if (FAILED(hr)) {
printf("IAudioClient::GetService(IAudioCaptureClient) failed: hr 0x%08x\n", hr);
audioClient->Release();
return hr;
}
hr = audioClient->Start();
if (FAILED(hr)) {
printf("IAudioClient::Start failed: hr = 0x%08x\n", hr);
audioCaptureClient->Release();
audioClient->Release();
return hr;
}
// loopback capture loop
for (UINT32 i = 0; true; i++) {
UINT32 nextPacketSize;
hr = audioCaptureClient->GetNextPacketSize(&nextPacketSize);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetNextPacketSize failed on pass %u: hr = 0x%08x\n", i, hr);
audioClient->Stop();
audioCaptureClient->Release();
audioClient->Release();
return hr;
}
if (nextPacketSize == 0) { // no data yet
continue;
}
// get the captured data
BYTE *data;
UINT32 frameCount;
DWORD bufferFlags;
hr = audioCaptureClient->GetBuffer(&data, &frameCount, &bufferFlags, NULL, NULL);
if (FAILED(hr)) {
printf("IAudioCaptureClient::GetBuffer failed on pass %u: hr = 0x%08x\n", i, hr);
audioClient->Stop();
audioCaptureClient->Release();
audioClient->Release();
return hr;
}
if (bufferFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) {
printf("IAudioCaptureClient::GetBuffer reports 'data discontinuity' on pass %u\n", i);
}
if (bufferFlags & AUDCLNT_BUFFERFLAGS_SILENT) {
printf("IAudioCaptureClient::GetBuffer reports 'silent' on pass %u\n", i);
}
if (bufferFlags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR) {
printf("IAudioCaptureClient::GetBuffer reports 'timestamp error' on pass %u\n", i);
}
if (frameCount == 0) {
printf("IAudioCaptureClient::GetBuffer said to read 0 frames on pass %u\n", i);
audioClient->Stop();
audioCaptureClient->Release();
audioClient->Release();
return E_UNEXPECTED;
}
LONG bytesToWrite = frameCount * blockAlign;
DWORD bytesWritten;
printf("Recording:%d\n",bytesToWrite);
fwrite(data,bytesToWrite,1,f);
/*WriteFile(processInWrite,reinterpret_cast<PCHAR>(data), bytesToWrite, &bytesWritten, NULL);
if (bytesWritten != bytesToWrite) {
printf("WriteFile: tried to write %d bytes, but %d bytes written\n", bytesToWrite, bytesWritten);
}
char buf[10000];
DWORD count;
DWORD bytesAvailable;
PeekNamedPipe(processOutRead, NULL, 0, 0, &bytesAvailable, NULL);
if (bytesAvailable > 0) {
ReadFile(processOutRead, buf, 10000, &count, NULL);
std::cout.write(buf, count);
}*/
hr = audioCaptureClient->ReleaseBuffer(frameCount);
if (FAILED(hr)) {
printf("IAudioCaptureClient::ReleaseBuffer failed on pass %u: hr = 0x%08x\n", i, hr);
audioClient->Stop();
audioCaptureClient->Release();
audioClient->Release();
return hr;
}
} // capture loop
audioClient->Stop();
audioCaptureClient->Release();
audioClient->Release();
fclose(f);
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;
}
