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"));
}
HRESULT __stdcall initialize_patch(
IAudioClient* this_, AUDCLNT_SHAREMODE ShareMode, DWORD StreamFlags,
REFERENCE_TIME hnsBufferDuration, REFERENCE_TIME hnsPeriodicity,
const WAVEFORMATEX* pFormat, LPCGUID AudioSessionGuid)
{
// synchronize initializing so it doesn't happen while streams are being flushed
HANDLE audio_router_mutex = OpenMutexW(SYNCHRONIZE, FALSE, L"Local\\audio-router-mutex");
assert(audio_router_mutex != NULL);
if(audio_router_mutex)
{
DWORD res = WaitForSingleObject(audio_router_mutex, INFINITE);
assert(res == WAIT_OBJECT_0);
}
IAudioClient* proxy = get_duplicate(this_)->proxy;
LPCGUID guid = ((GUID***)this_)[0][17];
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->Initialize(
ShareMode,
StreamFlags |
AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED |
AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED,
hnsBufferDuration,
hnsPeriodicity,
pFormat,
guid);
((DWORD_PTR**)this_)[0] = old_vftptr;
if(hr != S_OK)
tell_error(hr);
else
*((WORD***)this_)[0][18] = pFormat->nBlockAlign;
if(hr == S_OK)
{
for(iaudioclient_duplicate* next = get_duplicate(this_)->next;
next != NULL; next = next->next)
{
HRESULT hr2 = next->proxy->Initialize(
ShareMode,
StreamFlags |
AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM |
AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY |
AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED |
AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED,
hnsBufferDuration,
hnsPeriodicity,
pFormat,
guid);
if(hr2 != S_OK)
tell_error(hr2);
}
}
ReleaseMutex(audio_router_mutex);
CloseHandle(audio_router_mutex);
return hr;
}
HRESULT __stdcall getmixformat_patch(IAudioClient* this_, WAVEFORMATEX** ppDeviceFormat)
{
// STATIC FUNCTION
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetMixFormat(ppDeviceFormat);
((DWORD_PTR**)this_)[0] = old_vftptr;
return hr;
}
HRESULT __stdcall reset_patch(IAudioClient* this_)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->Reset();
((DWORD_PTR**)this_)[0] = old_vftptr;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
next->proxy->Reset();
return hr;
}
HRESULT __stdcall seteventhandle_patch(IAudioClient* this_, HANDLE eventHandle)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
next->proxy->SetEventHandle(eventHandle);
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->SetEventHandle(eventHandle);
((DWORD_PTR**)this_)[0] = old_vftptr;
return hr;
}
HRESULT __stdcall isformatsupported_patch(
IAudioClient* this_, AUDCLNT_SHAREMODE ShareMode,
const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch)
{
// STATIC FUNCTION
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->IsFormatSupported(ShareMode, pFormat, ppClosestMatch);
((DWORD_PTR**)this_)[0] = old_vftptr;
return hr;
}
HRESULT __stdcall start_patch(IAudioClient* this_)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->Start(), hr2;
((DWORD_PTR**)this_)[0] = old_vftptr;
if(hr == S_OK)
{
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
hr2 = next->proxy->Start();
}
return hr;
}
HRESULT __stdcall getbuffersize_patch(IAudioClient* this_, UINT32* pNumBufferFrames)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetBufferSize(pNumBufferFrames);
((DWORD_PTR**)this_)[0] = old_vftptr;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
{
UINT32 buf;
HRESULT hr = next->proxy->GetBufferSize(&buf);
assert(buf >= *pNumBufferFrames);
}
return hr;
}
HRESULT __stdcall getstreamlatency_patch(IAudioClient* this_, REFERENCE_TIME* phnsLatency)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetStreamLatency(phnsLatency);
((DWORD_PTR**)this_)[0] = old_vftptr;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
{
REFERENCE_TIME t;
next->proxy->GetStreamLatency(&t);
assert(*phnsLatency == t);
}
return hr;
}
HRESULT __stdcall getcurrentpadding_patch(IAudioClient* this_, UINT32* pNumPaddingFrames)
{
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetCurrentPadding(pNumPaddingFrames);
((DWORD_PTR**)this_)[0] = old_vftptr;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
{
UINT32 pad;
next->proxy->GetCurrentPadding(&pad);
//assert(pad == *pNumPaddingFrames);
}
return hr;
}
//
// 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;
}
//
// Retrieve the format we'll use to capture samples.
//
// We use the Mix format since we're capturing in shared mode.
//
bool CWASAPICapture::LoadFormat()
{
HRESULT hr = _AudioClient->GetMixFormat((WAVEFORMATEX**)&_MixFormat);
PersistentAssert(SUCCEEDED(hr), "_AudioClient->GetMixFormat failed");
//- SubFormat {00000003-0000-0010-8000-00AA00389B71} _GUID, KSDATAFORMAT_SUBTYPE_IEEE_FLOAT
_FrameSize = (_MixFormat->Format.wBitsPerSample / 8) * _MixFormat->Format.nChannels;
return true;
}
~BlankAudioPlayback()
{
mmClient->Stop();
SafeRelease(mmRender);
SafeRelease(mmClient);
SafeRelease(mmDevice);
SafeRelease(mmEnumerator);
}
HRESULT __stdcall getdeviceperiod_patch(
IAudioClient* this_,
REFERENCE_TIME* phnsDefaultDevicePeriod, REFERENCE_TIME* phnsMinimumDevicePeriod)
{
// STATIC FUNCTION
IAudioClient* proxy = get_duplicate(this_)->proxy;
DWORD_PTR* old_vftptr = swap_vtable(this_);
HRESULT hr = proxy->GetDevicePeriod(phnsDefaultDevicePeriod, phnsMinimumDevicePeriod);
((DWORD_PTR**)this_)[0] = old_vftptr;
for(iaudioclient_duplicate* next = get_duplicate(this_)->next; next != NULL; next = next->next)
{
REFERENCE_TIME def, min;
next->proxy->GetDevicePeriod(&def, &min);
assert(def == *phnsDefaultDevicePeriod && min == *phnsMinimumDevicePeriod);
}
return hr;
}
void MMDeviceAudioSource::StartCapture()
{
if(mmClient) {
mmClient->Start();
UINT64 freq;
mmClock->GetFrequency(&freq);
Log(TEXT("MMDeviceAudioSource: Frequency for device '%s' is %llu, samples per sec is %u"), GetDeviceName(), freq, this->GetSamplesPerSec());
}
}
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;
}
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);
}
//
// Start capturing...
//
bool CWASAPICapture::Start(BYTE *CaptureBuffer, size_t CaptureBufferSize)
{
HRESULT hr;
_CaptureBuffer = CaptureBuffer;
_CaptureBufferSize = CaptureBufferSize;
//
// Now create the thread which is going to drive the capture.
//
_CaptureThread = CreateThread(NULL, 0, WASAPICaptureThread, this, 0, NULL);
PersistentAssert(_CaptureThread != NULL, "CreateThread failed");
//
// We're ready to go, start capturing!
//
hr = _AudioClient->Start();
PersistentAssert(SUCCEEDED(hr), "_AudioClient->Start failed");
return true;
}
//
// Stop the capturer.
//
void CWASAPICapture::Stop()
{
HRESULT hr;
//
// Tell the capture thread to shut down, wait for the thread to complete then clean up all the stuff we
// allocated in Start().
//
if (_ShutdownEvent)
{
SetEvent(_ShutdownEvent);
}
hr = _AudioClient->Stop();
PersistentAssert(SUCCEEDED(hr), "_AudioClient->Stop failed");
if (_CaptureThread)
{
WaitForSingleObject(_CaptureThread, INFINITE);
CloseHandle(_CaptureThread);
_CaptureThread = NULL;
}
}
//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;
}
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);
}
void MMDeviceAudioSource::StopCapture()
{
if(mmClient)
mmClient->Stop();
}
void CAESinkDirectSound::EnumerateDevicesEx(AEDeviceInfoList &deviceInfoList, bool force)
{
CAEDeviceInfo deviceInfo;
IMMDeviceEnumerator* pEnumerator = NULL;
IMMDeviceCollection* pEnumDevices = NULL;
HRESULT hr;
std::string strDD = GetDefaultDevice();
/* Windows Vista or later - supporting WASAPI device probing */
hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&pEnumerator);
EXIT_ON_FAILURE(hr, __FUNCTION__": Could not allocate WASAPI device enumerator. CoCreateInstance error code: %li", hr)
UINT uiCount = 0;
hr = pEnumerator->EnumAudioEndpoints(eRender, DEVICE_STATE_ACTIVE, &pEnumDevices);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint enumeration failed.")
hr = pEnumDevices->GetCount(&uiCount);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint count failed.")
for (UINT i = 0; i < uiCount; i++)
{
IMMDevice *pDevice = NULL;
IPropertyStore *pProperty = NULL;
PROPVARIANT varName;
PropVariantInit(&varName);
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
hr = pEnumDevices->Item(i, &pDevice);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint failed.");
goto failed;
}
hr = pDevice->OpenPropertyStore(STGM_READ, &pProperty);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint properties failed.");
SAFE_RELEASE(pDevice);
goto failed;
}
hr = pProperty->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint device name failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strFriendlyName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_GUID, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint GUID failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strDevName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_FormFactor, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint form factor failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strWinDevType = winEndpoints[(EndpointFormFactor)varName.uiVal].winEndpointType;
AEDeviceType aeDeviceType = winEndpoints[(EndpointFormFactor)varName.uiVal].aeDeviceType;
PropVariantClear(&varName);
/* In shared mode Windows tells us what format the audio must be in. */
IAudioClient *pClient;
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pClient);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Activate device failed (%s)", WASAPIErrToStr(hr));
goto failed;
}
//hr = pClient->GetMixFormat(&pwfxex);
hr = pProperty->GetValue(PKEY_AudioEngine_DeviceFormat, &varName);
if (SUCCEEDED(hr) && varName.blob.cbSize > 0)
{
WAVEFORMATEX* smpwfxex = (WAVEFORMATEX*)varName.blob.pBlobData;
deviceInfo.m_channels = layoutsByChCount[std::max(std::min(smpwfxex->nChannels, (WORD) DS_SPEAKER_COUNT), (WORD) 2)];
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_FLOAT));
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
deviceInfo.m_sampleRates.push_back(std::min(smpwfxex->nSamplesPerSec, (DWORD) 192000));
}
else
{
CLog::Log(LOGERROR, __FUNCTION__": Getting DeviceFormat failed (%s)", WASAPIErrToStr(hr));
}
pClient->Release();
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
deviceInfo.m_deviceName = strDevName;
deviceInfo.m_displayName = strWinDevType.append(strFriendlyName);
deviceInfo.m_displayNameExtra = std::string("DirectSound: ").append(strFriendlyName);
deviceInfo.m_deviceType = aeDeviceType;
deviceInfoList.push_back(deviceInfo);
// add the default device with m_deviceName = default
if(strDD == strDevName)
{
deviceInfo.m_deviceName = std::string("default");
deviceInfo.m_displayName = std::string("default");
deviceInfo.m_displayNameExtra = std::string("");
deviceInfoList.push_back(deviceInfo);
}
}
return;
failed:
if (FAILED(hr))
CLog::Log(LOGERROR, __FUNCTION__": Failed to enumerate WASAPI endpoint devices (%s).", WASAPIErrToStr(hr));
SAFE_RELEASE(pEnumDevices);
SAFE_RELEASE(pEnumerator);
}
void CAESinkDirectSound::EnumerateDevicesEx(AEDeviceInfoList &deviceInfoList)
{
CAEDeviceInfo deviceInfo;
OSVERSIONINFO osvi;
IMMDeviceEnumerator* pEnumerator = NULL;
IMMDeviceCollection* pEnumDevices = NULL;
WAVEFORMATEX* pwfxex = NULL;
HRESULT hr;
/* See if we are on Windows XP */
ZeroMemory(&osvi, sizeof(OSVERSIONINFO));
osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
GetVersionEx(&osvi);
if (osvi.dwMajorVersion == 5)
{
/* We are on XP - WASAPI not supported - enumerate using DS devices */
LPGUID deviceGUID = NULL;
RPC_CSTR cszGUID;
std::string szGUID;
std::list<DSDevice> DSDeviceList;
DirectSoundEnumerate(DSEnumCallback, &DSDeviceList);
for(std::list<DSDevice>::iterator itt = DSDeviceList.begin(); itt != DSDeviceList.end(); itt++)
{
if (UuidToString((*itt).lpGuid, &cszGUID) != RPC_S_OK)
continue; /* could not convert GUID to string - skip device */
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
szGUID = (LPSTR)cszGUID;
deviceInfo.m_deviceName = "{" + szGUID + "}";
deviceInfo.m_displayName = (*itt).name;
deviceInfo.m_displayNameExtra = std::string("DirectSound: ") + (*itt).name;
deviceInfo.m_deviceType = AE_DEVTYPE_PCM;
deviceInfo.m_channels = layoutsByChCount[2];
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_FLOAT));
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
deviceInfo.m_sampleRates.push_back((DWORD) 96000);
deviceInfoList.push_back(deviceInfo);
}
RpcStringFree(&cszGUID);
return;
}
/* Windows Vista or later - supporting WASAPI device probing */
hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&pEnumerator);
EXIT_ON_FAILURE(hr, __FUNCTION__": Could not allocate WASAPI device enumerator. CoCreateInstance error code: %li", hr)
UINT uiCount = 0;
hr = pEnumerator->EnumAudioEndpoints(eRender, DEVICE_STATE_ACTIVE, &pEnumDevices);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint enumeration failed.")
hr = pEnumDevices->GetCount(&uiCount);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint count failed.")
for (UINT i = 0; i < uiCount; i++)
{
IMMDevice *pDevice = NULL;
IPropertyStore *pProperty = NULL;
PROPVARIANT varName;
PropVariantInit(&varName);
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
hr = pEnumDevices->Item(i, &pDevice);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint failed.");
goto failed;
}
hr = pDevice->OpenPropertyStore(STGM_READ, &pProperty);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint properties failed.");
SAFE_RELEASE(pDevice);
goto failed;
}
hr = pProperty->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint device name failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::wstring strRawFriendlyName(varName.pwszVal);
std::string strFriendlyName = std::string(strRawFriendlyName.begin(), strRawFriendlyName.end());
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_GUID, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint GUID failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::wstring strRawDevName(varName.pwszVal);
std::string strDevName = std::string(strRawDevName.begin(), strRawDevName.end());
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_FormFactor, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint form factor failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strWinDevType = winEndpoints[(EndpointFormFactor)varName.uiVal].winEndpointType;
AEDeviceType aeDeviceType = winEndpoints[(EndpointFormFactor)varName.uiVal].aeDeviceType;
PropVariantClear(&varName);
/* In shared mode Windows tells us what format the audio must be in. */
IAudioClient *pClient;
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pClient);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Activate device failed (%s)", WASAPIErrToStr(hr));
}
//hr = pClient->GetMixFormat(&pwfxex);
hr = pProperty->GetValue(PKEY_AudioEngine_DeviceFormat, &varName);
if (SUCCEEDED(hr))
{
WAVEFORMATEX* smpwfxex = (WAVEFORMATEX*)varName.blob.pBlobData;
deviceInfo.m_channels = layoutsByChCount[std::min(smpwfxex->nChannels, (WORD) 2)];
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_FLOAT));
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
deviceInfo.m_sampleRates.push_back(std::min(smpwfxex->nSamplesPerSec, (DWORD) 96000));
}
else
{
CLog::Log(LOGERROR, __FUNCTION__": GetMixFormat failed (%s)", WASAPIErrToStr(hr));
}
pClient->Release();
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
deviceInfo.m_deviceName = strDevName;
deviceInfo.m_displayName = strWinDevType.append(strFriendlyName);
deviceInfo.m_displayNameExtra = std::string("DirectSound: ").append(strFriendlyName);
deviceInfo.m_deviceType = aeDeviceType;
/* Now logged by AESinkFactory on startup */
//CLog::Log(LOGDEBUG,"Audio Device %d: %s", i, ((std::string)deviceInfo).c_str());
deviceInfoList.push_back(deviceInfo);
}
return;
failed:
if (FAILED(hr))
CLog::Log(LOGERROR, __FUNCTION__": Failed to enumerate WASAPI endpoint devices (%s).", WASAPIErrToStr(hr));
SAFE_RELEASE(pEnumDevices);
SAFE_RELEASE(pEnumerator);
}
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 CAESinkDirectSound::EnumerateDevicesEx(AEDeviceInfoList &deviceInfoList, bool force)
{
CAEDeviceInfo deviceInfo;
IMMDeviceEnumerator* pEnumerator = NULL;
IMMDeviceCollection* pEnumDevices = NULL;
HRESULT hr;
/* See if we are on Windows XP */
if (!g_sysinfo.IsWindowsVersionAtLeast(CSysInfo::WindowsVersionVista))
{
/* We are on XP - WASAPI not supported - enumerate using DS devices */
LPGUID deviceGUID = NULL;
RPC_CSTR cszGUID;
std::string szGUID;
std::list<DSDevice> DSDeviceList;
DirectSoundEnumerate(DSEnumCallback, &DSDeviceList);
for(std::list<DSDevice>::iterator itt = DSDeviceList.begin(); itt != DSDeviceList.end(); ++itt)
{
if (UuidToString((*itt).lpGuid, &cszGUID) != RPC_S_OK)
continue; /* could not convert GUID to string - skip device */
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
szGUID = (LPSTR)cszGUID;
deviceInfo.m_deviceName = "{" + szGUID + "}";
deviceInfo.m_displayName = (*itt).name;
deviceInfo.m_displayNameExtra = std::string("DirectSound: ") + (*itt).name;
deviceInfo.m_deviceType = AE_DEVTYPE_PCM;
deviceInfo.m_channels = layoutsByChCount[2];
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_FLOAT));
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
deviceInfo.m_sampleRates.push_back((DWORD) 96000);
deviceInfoList.push_back(deviceInfo);
}
RpcStringFree(&cszGUID);
return;
}
/* Windows Vista or later - supporting WASAPI device probing */
hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&pEnumerator);
EXIT_ON_FAILURE(hr, __FUNCTION__": Could not allocate WASAPI device enumerator. CoCreateInstance error code: %li", hr)
UINT uiCount = 0;
hr = pEnumerator->EnumAudioEndpoints(eRender, DEVICE_STATE_ACTIVE, &pEnumDevices);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint enumeration failed.")
hr = pEnumDevices->GetCount(&uiCount);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint count failed.")
for (UINT i = 0; i < uiCount; i++)
{
IMMDevice *pDevice = NULL;
IPropertyStore *pProperty = NULL;
PROPVARIANT varName;
PropVariantInit(&varName);
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
hr = pEnumDevices->Item(i, &pDevice);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint failed.");
goto failed;
}
hr = pDevice->OpenPropertyStore(STGM_READ, &pProperty);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint properties failed.");
SAFE_RELEASE(pDevice);
goto failed;
}
hr = pProperty->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint device name failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strFriendlyName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_GUID, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint GUID failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strDevName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_FormFactor, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of DirectSound endpoint form factor failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strWinDevType = winEndpoints[(EndpointFormFactor)varName.uiVal].winEndpointType;
AEDeviceType aeDeviceType = winEndpoints[(EndpointFormFactor)varName.uiVal].aeDeviceType;
PropVariantClear(&varName);
/* In shared mode Windows tells us what format the audio must be in. */
IAudioClient *pClient;
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pClient);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Activate device failed (%s)", WASAPIErrToStr(hr));
goto failed;
}
//hr = pClient->GetMixFormat(&pwfxex);
hr = pProperty->GetValue(PKEY_AudioEngine_DeviceFormat, &varName);
if (SUCCEEDED(hr) && varName.blob.cbSize > 0)
{
WAVEFORMATEX* smpwfxex = (WAVEFORMATEX*)varName.blob.pBlobData;
deviceInfo.m_channels = layoutsByChCount[std::max(std::min(smpwfxex->nChannels, (WORD) DS_SPEAKER_COUNT), (WORD) 2)];
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_FLOAT));
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
deviceInfo.m_sampleRates.push_back(std::min(smpwfxex->nSamplesPerSec, (DWORD) 192000));
}
else
{
CLog::Log(LOGERROR, __FUNCTION__": Getting DeviceFormat failed (%s)", WASAPIErrToStr(hr));
}
pClient->Release();
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
deviceInfo.m_deviceName = strDevName;
deviceInfo.m_displayName = strWinDevType.append(strFriendlyName);
deviceInfo.m_displayNameExtra = std::string("DirectSound: ").append(strFriendlyName);
deviceInfo.m_deviceType = aeDeviceType;
deviceInfoList.push_back(deviceInfo);
}
// since AE takes the first device in deviceInfoList as default audio device we need
// to sort it in order to use the real default device
if(deviceInfoList.size() > 1)
{
std::string strDD = GetDefaultDevice();
for (AEDeviceInfoList::iterator itt = deviceInfoList.begin(); itt != deviceInfoList.end(); ++itt)
{
CAEDeviceInfo devInfo = *itt;
if(devInfo.m_deviceName == strDD)
{
deviceInfoList.erase(itt);
deviceInfoList.insert(deviceInfoList.begin(), devInfo);
break;
}
}
}
return;
failed:
if (FAILED(hr))
CLog::Log(LOGERROR, __FUNCTION__": Failed to enumerate WASAPI endpoint devices (%s).", WASAPIErrToStr(hr));
SAFE_RELEASE(pEnumDevices);
SAFE_RELEASE(pEnumerator);
}
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);
}
void CAESinkWASAPI::EnumerateDevicesEx(AEDeviceInfoList &deviceInfoList, bool force)
{
IMMDeviceEnumerator* pEnumerator = NULL;
IMMDeviceCollection* pEnumDevices = NULL;
IMMDevice* pDefaultDevice = NULL;
CAEDeviceInfo deviceInfo;
CAEChannelInfo deviceChannels;
LPWSTR pwszID = NULL;
std::wstring wstrDDID;
WAVEFORMATEXTENSIBLE wfxex = {0};
HRESULT hr;
hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, IID_IMMDeviceEnumerator, (void**)&pEnumerator);
EXIT_ON_FAILURE(hr, __FUNCTION__": Could not allocate WASAPI device enumerator. CoCreateInstance error code: %li", hr)
UINT uiCount = 0;
// get the default audio endpoint
if(pEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &pDefaultDevice) == S_OK)
{
if(pDefaultDevice->GetId(&pwszID) == S_OK)
{
wstrDDID = pwszID;
CoTaskMemFree(pwszID);
}
SAFE_RELEASE(pDefaultDevice);
}
// enumerate over all audio endpoints
hr = pEnumerator->EnumAudioEndpoints(eRender, DEVICE_STATE_ACTIVE, &pEnumDevices);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint enumeration failed.")
hr = pEnumDevices->GetCount(&uiCount);
EXIT_ON_FAILURE(hr, __FUNCTION__": Retrieval of audio endpoint count failed.")
for (UINT i = 0; i < uiCount; i++)
{
IMMDevice *pDevice = NULL;
IPropertyStore *pProperty = NULL;
PROPVARIANT varName;
PropVariantInit(&varName);
deviceInfo.m_channels.Reset();
deviceInfo.m_dataFormats.clear();
deviceInfo.m_sampleRates.clear();
hr = pEnumDevices->Item(i, &pDevice);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint failed.");
goto failed;
}
hr = pDevice->OpenPropertyStore(STGM_READ, &pProperty);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint properties failed.");
SAFE_RELEASE(pDevice);
goto failed;
}
hr = pProperty->GetValue(PKEY_Device_FriendlyName, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint device name failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strFriendlyName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_GUID, &varName);
if(FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint GUID failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strDevName = localWideToUtf(varName.pwszVal);
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_FormFactor, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint form factor failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
std::string strWinDevType = winEndpoints[(EndpointFormFactor)varName.uiVal].winEndpointType;
AEDeviceType aeDeviceType = winEndpoints[(EndpointFormFactor)varName.uiVal].aeDeviceType;
PropVariantClear(&varName);
hr = pProperty->GetValue(PKEY_AudioEndpoint_PhysicalSpeakers, &varName);
if (FAILED(hr))
{
CLog::Log(LOGERROR, __FUNCTION__": Retrieval of WASAPI endpoint speaker layout failed.");
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
goto failed;
}
unsigned int uiChannelMask = std::max(varName.uintVal, (unsigned int) KSAUDIO_SPEAKER_STEREO);
deviceChannels.Reset();
for (unsigned int c = 0; c < WASAPI_SPEAKER_COUNT; c++)
{
if (uiChannelMask & WASAPIChannelOrder[c])
deviceChannels += AEChannelNames[c];
}
PropVariantClear(&varName);
IAudioClient *pClient;
hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pClient);
if (SUCCEEDED(hr))
{
/* Test format DTS-HD */
wfxex.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
wfxex.Format.nSamplesPerSec = 192000;
wfxex.dwChannelMask = KSAUDIO_SPEAKER_7POINT1_SURROUND;
wfxex.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DTS_HD;
wfxex.Format.wBitsPerSample = 16;
wfxex.Samples.wValidBitsPerSample = 16;
wfxex.Format.nChannels = 8;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_DTSHD));
/* Test format Dolby TrueHD */
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_MLP;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_TRUEHD));
/* Test format Dolby EAC3 */
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL_PLUS;
wfxex.Format.nChannels = 2;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_EAC3));
/* Test format DTS */
wfxex.Format.nSamplesPerSec = 48000;
wfxex.dwChannelMask = KSAUDIO_SPEAKER_5POINT1;
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DTS;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_DTS));
/* Test format Dolby AC3 */
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_DOLBY_DIGITAL;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AC3));
/* Test format AAC */
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEC61937_AAC;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back(AEDataFormat(AE_FMT_AAC));
/* Test format for PCM format iteration */
wfxex.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
wfxex.dwChannelMask = KSAUDIO_SPEAKER_STEREO;
wfxex.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
for (int p = AE_FMT_FLOAT; p > AE_FMT_INVALID; p--)
{
if (p < AE_FMT_FLOAT)
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wfxex.Format.wBitsPerSample = CAEUtil::DataFormatToBits((AEDataFormat) p);
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
if (p <= AE_FMT_S24NE4 && p >= AE_FMT_S24BE4)
{
wfxex.Samples.wValidBitsPerSample = 24;
}
else
{
wfxex.Samples.wValidBitsPerSample = wfxex.Format.wBitsPerSample;
}
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_dataFormats.push_back((AEDataFormat) p);
}
/* Test format for sample rate iteration */
wfxex.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
wfxex.dwChannelMask = KSAUDIO_SPEAKER_STEREO;
wfxex.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wfxex.Format.wBitsPerSample = 16;
wfxex.Samples.wValidBitsPerSample = 16;
wfxex.Format.nChannels = 2;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
for (int j = 0; j < WASAPISampleRateCount; j++)
{
wfxex.Format.nSamplesPerSec = WASAPISampleRates[j];
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
deviceInfo.m_sampleRates.push_back(WASAPISampleRates[j]);
}
/* Test format for channels iteration */
wfxex.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
wfxex.dwChannelMask = KSAUDIO_SPEAKER_STEREO;
wfxex.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
wfxex.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wfxex.Format.nSamplesPerSec = 48000;
wfxex.Format.wBitsPerSample = 16;
wfxex.Samples.wValidBitsPerSample = 16;
wfxex.Format.nChannels = 2;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
bool hasLpcm = false;
// Try with KSAUDIO_SPEAKER_DIRECTOUT
for (unsigned int k = WASAPI_SPEAKER_COUNT; k > 0; k--)
{
wfxex.dwChannelMask = KSAUDIO_SPEAKER_DIRECTOUT;
wfxex.Format.nChannels = k;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
{
if (k > 3) // Add only multichannel LPCM
{
deviceInfo.m_dataFormats.push_back(AE_FMT_LPCM);
hasLpcm = true;
}
break;
}
}
/* Try with reported channel mask */
for (unsigned int k = WASAPI_SPEAKER_COUNT; k > 0; k--)
{
wfxex.dwChannelMask = uiChannelMask;
wfxex.Format.nChannels = k;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
{
if ( !hasLpcm && k > 3) // Add only multichannel LPCM
{
deviceInfo.m_dataFormats.push_back(AE_FMT_LPCM);
hasLpcm = true;
}
break;
}
}
/* Try with specific speakers configurations */
for (unsigned int i = 0; i < ARRAYSIZE(layoutsList); i++)
{
unsigned int nmbOfCh;
wfxex.dwChannelMask = ChLayoutToChMask(layoutsList[i], &nmbOfCh);
wfxex.Format.nChannels = nmbOfCh;
wfxex.Format.nBlockAlign = wfxex.Format.nChannels * (wfxex.Format.wBitsPerSample >> 3);
wfxex.Format.nAvgBytesPerSec = wfxex.Format.nSamplesPerSec * wfxex.Format.nBlockAlign;
hr = pClient->IsFormatSupported(AUDCLNT_SHAREMODE_EXCLUSIVE, &wfxex.Format, NULL);
if (SUCCEEDED(hr))
{
if ( deviceChannels.Count() < nmbOfCh)
deviceChannels = layoutsList[i];
if ( !hasLpcm && nmbOfCh > 3) // Add only multichannel LPCM
{
deviceInfo.m_dataFormats.push_back(AE_FMT_LPCM);
hasLpcm = true;
}
}
}
pClient->Release();
}
else
{
CLog::Log(LOGDEBUG, __FUNCTION__": Failed to activate device for passthrough capability testing.");
}
deviceInfo.m_deviceName = strDevName;
deviceInfo.m_displayName = strWinDevType.append(strFriendlyName);
deviceInfo.m_displayNameExtra = std::string("WASAPI: ").append(strFriendlyName);
deviceInfo.m_deviceType = aeDeviceType;
deviceInfo.m_channels = deviceChannels;
/* Store the device info */
deviceInfoList.push_back(deviceInfo);
if(pDevice->GetId(&pwszID) == S_OK)
{
if(wstrDDID.compare(pwszID) == 0)
{
deviceInfo.m_deviceName = std::string("default");
deviceInfo.m_displayName = std::string("default");
deviceInfo.m_displayNameExtra = std::string("");
deviceInfoList.push_back(deviceInfo);
}
CoTaskMemFree(pwszID);
}
SAFE_RELEASE(pDevice);
SAFE_RELEASE(pProperty);
}
