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 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;
}
//
// 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;
}
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);
}
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;
}
