static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, DWORD dwUser, DWORD dw1, DWORD dw2)
{
internalPortAudioStream *past;
PaHostSoundControl *pahsc;
#if PA_SIMULATE_UNDERFLOW
gUnderCallbackCounter++;
if( (gUnderCallbackCounter >= UNDER_START_GAP) &&
(gUnderCallbackCounter <= UNDER_STOP_GAP) )
{
if( gUnderCallbackCounter == UNDER_START_GAP)
{
AddTraceMessage("Begin stall: gUnderCallbackCounter =======", gUnderCallbackCounter );
}
if( gUnderCallbackCounter == UNDER_STOP_GAP)
{
AddTraceMessage("End stall: gUnderCallbackCounter =======", gUnderCallbackCounter );
}
return;
}
#endif
past = (internalPortAudioStream *) dwUser;
if( past == NULL ) return;
pahsc = (PaHostSoundControl *) past->past_DeviceData;
if( pahsc == NULL ) return;
if( !pahsc->pahsc_IfInsideCallback && past->past_IsActive )
{
if( past->past_StopNow )
{
past->past_IsActive = 0;
}
else if( past->past_StopSoon )
{
DSoundWrapper *dsw = &pahsc->pahsc_DSoundWrapper;
if( past->past_NumOutputChannels > 0 )
{
DSW_ZeroEmptySpace( dsw );
AddTraceMessage("Pa_TimerCallback: waiting - written ", (int) dsw->dsw_FramesWritten );
AddTraceMessage("Pa_TimerCallback: waiting - played ", (int) dsw->dsw_FramesPlayed );
/* clear past_IsActive when all sound played */
if( dsw->dsw_FramesPlayed >= past->past_FrameCount )
{
past->past_IsActive = 0;
}
}
else
{
past->past_IsActive = 0;
}
}
else
{
pahsc->pahsc_IfInsideCallback = 1;
if( Pa_TimeSlice( past ) != 0) /* Call time slice independant of timing method. */
{
past->past_StopSoon = 1;
}
pahsc->pahsc_IfInsideCallback = 0;
}
}
}
HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, int nChannels, int bytesPerBuffer )
{
DWORD dwDataLen;
DWORD playCursor;
HRESULT result;
LPDIRECTSOUNDBUFFER pPrimaryBuffer;
HWND hWnd;
HRESULT hr;
WAVEFORMATEX wfFormat;
DSBUFFERDESC primaryDesc;
DSBUFFERDESC secondaryDesc;
unsigned char* pDSBuffData;
LARGE_INTEGER counterFrequency;
dsw->dsw_OutputSize = bytesPerBuffer;
dsw->dsw_OutputRunning = FALSE;
dsw->dsw_OutputUnderflows = 0;
dsw->dsw_FramesWritten = 0;
dsw->dsw_BytesPerFrame = nChannels * sizeof(short);
// We were using getForegroundWindow() but sometimes the ForegroundWindow may not be the
// applications's window. Also if that window is closed before the Buffer is closed
// then DirectSound can crash. (Thanks for Scott Patterson for reporting this.)
// So we will use GetDesktopWindow() which was suggested by Miller Puckette.
// hWnd = GetForegroundWindow();
hWnd = GetDesktopWindow();
// Set cooperative level to DSSCL_EXCLUSIVE so that we can get 16 bit output, 44.1 KHz.
// Exclusize also prevents unexpected sounds from other apps during a performance.
if ((hr = IDirectSound_SetCooperativeLevel( dsw->dsw_pDirectSound,
hWnd, DSSCL_EXCLUSIVE)) != DS_OK)
{
return hr;
}
// -----------------------------------------------------------------------
// Create primary buffer and set format just so we can specify our custom format.
// Otherwise we would be stuck with the default which might be 8 bit or 22050 Hz.
// Setup the primary buffer description
ZeroMemory(&primaryDesc, sizeof(DSBUFFERDESC));
primaryDesc.dwSize = sizeof(DSBUFFERDESC);
primaryDesc.dwFlags = DSBCAPS_PRIMARYBUFFER; // all panning, mixing, etc done by synth
primaryDesc.dwBufferBytes = 0;
primaryDesc.lpwfxFormat = NULL;
// Create the buffer
if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound,
&primaryDesc, &pPrimaryBuffer, NULL)) != DS_OK) return result;
// Define the buffer format
wfFormat.wFormatTag = WAVE_FORMAT_PCM;
wfFormat.nChannels = nChannels;
wfFormat.nSamplesPerSec = nFrameRate;
wfFormat.wBitsPerSample = 8 * sizeof(short);
wfFormat.nBlockAlign = wfFormat.nChannels * wfFormat.wBitsPerSample / 8;
wfFormat.nAvgBytesPerSec = wfFormat.nSamplesPerSec * wfFormat.nBlockAlign;
wfFormat.cbSize = 0; /* No extended format info. */
// Set the primary buffer's format
if((result = IDirectSoundBuffer_SetFormat( pPrimaryBuffer, &wfFormat)) != DS_OK) return result;
// ----------------------------------------------------------------------
// Setup the secondary buffer description
ZeroMemory(&secondaryDesc, sizeof(DSBUFFERDESC));
secondaryDesc.dwSize = sizeof(DSBUFFERDESC);
secondaryDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
secondaryDesc.dwBufferBytes = bytesPerBuffer;
secondaryDesc.lpwfxFormat = &wfFormat;
// Create the secondary buffer
if ((result = IDirectSound_CreateSoundBuffer( dsw->dsw_pDirectSound,
&secondaryDesc, &dsw->dsw_OutputBuffer, NULL)) != DS_OK) return result;
// Lock the DS buffer
if ((result = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, 0, dsw->dsw_OutputSize, (LPVOID*)&pDSBuffData,
&dwDataLen, NULL, 0, 0)) != DS_OK) return result;
// Zero the DS buffer
ZeroMemory(pDSBuffData, dwDataLen);
// Unlock the DS buffer
if ((result = IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, pDSBuffData, dwDataLen, NULL, 0)) != DS_OK) return result;
if( QueryPerformanceFrequency( &counterFrequency ) )
{
int framesInBuffer = bytesPerBuffer / (nChannels * sizeof(short));
dsw->dsw_CounterTicksPerBuffer.QuadPart = (counterFrequency.QuadPart * framesInBuffer) / nFrameRate;
AddTraceMessage("dsw_CounterTicksPerBuffer = %d\n", dsw->dsw_CounterTicksPerBuffer.LowPart );
}
else
{
dsw->dsw_CounterTicksPerBuffer.QuadPart = 0;
}
// Let DSound set the starting write position because if we set it to zero, it looks like the
// buffer is full to begin with. This causes a long pause before sound starts when using large buffers.
hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &dsw->dsw_WriteOffset );
if( hr != DS_OK )
{
return hr;
}
dsw->dsw_FramesWritten = dsw->dsw_WriteOffset / dsw->dsw_BytesPerFrame;
/* printf("DSW_InitOutputBuffer: playCursor = %d, writeCursor = %d\n", playCursor, dsw->dsw_WriteOffset ); */
return DS_OK;
}
HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty )
{
HRESULT hr;
DWORD playCursor;
DWORD writeCursor;
long numBytesEmpty;
long playWriteGap;
// Query to see how much room is in buffer.
// Note: Even though writeCursor is not used, it must be passed to prevent DirectSound from dieing
// under WinNT. The Microsoft documentation says we can pass NULL but apparently not.
// Thanks to Max Rheiner for the fix.
hr = IDirectSoundBuffer_GetCurrentPosition( dsw->dsw_OutputBuffer, &playCursor, &writeCursor );
if( hr != DS_OK )
{
return hr;
}
AddTraceMessage("playCursor", playCursor);
AddTraceMessage("dsw_WriteOffset", dsw->dsw_WriteOffset);
// Determine size of gap between playIndex and WriteIndex that we cannot write into.
playWriteGap = writeCursor - playCursor;
if( playWriteGap < 0 ) playWriteGap += dsw->dsw_OutputSize; // unwrap
/* DirectSound doesn't have a large enough playCursor so we cannot detect wrap-around. */
/* Attempt to detect playCursor wrap-around and correct it. */
if( dsw->dsw_OutputRunning && (dsw->dsw_CounterTicksPerBuffer.QuadPart != 0) )
{
/* How much time has elapsed since last check. */
LARGE_INTEGER currentTime;
LARGE_INTEGER elapsedTime;
long bytesPlayed;
long bytesExpected;
long buffersWrapped;
QueryPerformanceCounter( ¤tTime );
elapsedTime.QuadPart = currentTime.QuadPart - dsw->dsw_LastPlayTime.QuadPart;
dsw->dsw_LastPlayTime = currentTime;
/* How many bytes does DirectSound say have been played. */
bytesPlayed = playCursor - dsw->dsw_LastPlayCursor;
if( bytesPlayed < 0 ) bytesPlayed += dsw->dsw_OutputSize; // unwrap
dsw->dsw_LastPlayCursor = playCursor;
/* Calculate how many bytes we would have expected to been played by now. */
bytesExpected = (long) ((elapsedTime.QuadPart * dsw->dsw_OutputSize) / dsw->dsw_CounterTicksPerBuffer.QuadPart);
buffersWrapped = (bytesExpected - bytesPlayed) / dsw->dsw_OutputSize;
if( buffersWrapped > 0 )
{
AddTraceMessage("playCursor wrapped! bytesPlayed", bytesPlayed );
AddTraceMessage("playCursor wrapped! bytesExpected", bytesExpected );
playCursor += (buffersWrapped * dsw->dsw_OutputSize);
bytesPlayed += (buffersWrapped * dsw->dsw_OutputSize);
}
/* Maintain frame output cursor. */
dsw->dsw_FramesPlayed += (bytesPlayed / dsw->dsw_BytesPerFrame);
}
numBytesEmpty = playCursor - dsw->dsw_WriteOffset;
if( numBytesEmpty < 0 ) numBytesEmpty += dsw->dsw_OutputSize; // unwrap offset
/* Have we underflowed? */
if( numBytesEmpty > (dsw->dsw_OutputSize - playWriteGap) )
{
if( dsw->dsw_OutputRunning )
{
dsw->dsw_OutputUnderflows += 1;
AddTraceMessage("underflow detected! numBytesEmpty", numBytesEmpty );
}
dsw->dsw_WriteOffset = writeCursor;
numBytesEmpty = dsw->dsw_OutputSize - playWriteGap;
}
*bytesEmpty = numBytesEmpty;
return hr;
}
static PaError Pa_TimeSlice( internalPortAudioStream *past )
{
PaError result = 0;
long bytesEmpty = 0;
long bytesFilled = 0;
long bytesToXfer = 0;
long numChunks;
HRESULT hresult;
PaHostSoundControl *pahsc;
short *nativeBufPtr;
past->past_NumCallbacks += 1;
pahsc = (PaHostSoundControl *) past->past_DeviceData;
if( pahsc == NULL ) return paInternalError;
/* How much input data is available? */
#if SUPPORT_AUDIO_CAPTURE
if( past->past_NumInputChannels > 0 )
{
DSW_QueryInputFilled( &pahsc->pahsc_DSoundWrapper, &bytesFilled );
bytesToXfer = bytesFilled;
}
#endif /* SUPPORT_AUDIO_CAPTURE */
/* How much output room is available? */
if( past->past_NumOutputChannels > 0 )
{
DSW_QueryOutputSpace( &pahsc->pahsc_DSoundWrapper, &bytesEmpty );
bytesToXfer = bytesEmpty;
}
AddTraceMessage( "bytesEmpty ", bytesEmpty );
/* Choose smallest value if both are active. */
if( (past->past_NumInputChannels > 0) && (past->past_NumOutputChannels > 0) )
{
bytesToXfer = ( bytesFilled < bytesEmpty ) ? bytesFilled : bytesEmpty;
}
/* printf("bytesFilled = %d, bytesEmpty = %d, bytesToXfer = %d\n",
bytesFilled, bytesEmpty, bytesToXfer);
*/
/* Quantize to multiples of a buffer. */
numChunks = bytesToXfer / pahsc->pahsc_BytesPerBuffer;
if( numChunks > (long)(past->past_NumUserBuffers/2) )
{
numChunks = (long)past->past_NumUserBuffers/2;
}
else if( numChunks < 0 )
{
numChunks = 0;
}
AddTraceMessage( "numChunks ", numChunks );
nativeBufPtr = pahsc->pahsc_NativeBuffer;
if( numChunks > 0 )
{
while( numChunks-- > 0 )
{
/* Measure usage based on time to process one user buffer. */
Pa_StartUsageCalculation( past );
#if SUPPORT_AUDIO_CAPTURE
/* Get native data from DirectSound. */
if( past->past_NumInputChannels > 0 )
{
hresult = DSW_ReadBlock( &pahsc->pahsc_DSoundWrapper, (char *) nativeBufPtr, pahsc->pahsc_BytesPerBuffer );
if( hresult < 0 )
{
ERR_RPT(("DirectSound ReadBlock failed, hresult = 0x%x\n",hresult));
sPaHostError = hresult;
break;
}
}
#endif /* SUPPORT_AUDIO_CAPTURE */
/* Convert 16 bit native data to user data and call user routine. */
result = Pa_CallConvertInt16( past, nativeBufPtr, nativeBufPtr );
if( result != 0) break;
/* Pass native data to DirectSound. */
if( past->past_NumOutputChannels > 0 )
{
/* static short DEBUGHACK = 0;
DEBUGHACK += 0x0049;
nativeBufPtr[0] = DEBUGHACK; /* Make buzz to see if DirectSound still running. */
hresult = DSW_WriteBlock( &pahsc->pahsc_DSoundWrapper, (char *) nativeBufPtr, pahsc->pahsc_BytesPerBuffer );
if( hresult < 0 )
{
ERR_RPT(("DirectSound WriteBlock failed, result = 0x%x\n",hresult));
sPaHostError = hresult;
break;
}
}
Pa_EndUsageCalculation( past );
}
}
return result;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** It may be called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int QaCallback( void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
PaTimestamp outTime, void *userData )
{
unsigned long i;
short phase;
PaQaData *data = (PaQaData *) userData;
(void) inputBuffer;
(void) outTime;
/* Play simple sawtooth wave. */
if( data->mode == MODE_OUTPUT )
{
phase = data->sawPhase;
switch( data->format )
{
case paFloat32:
{
float *out = (float *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = (float) (phase * (1.0 / 32768.0));
if( data->numChannels == 2 )
{
*out++ = (float) (phase * (1.0 / 32768.0));
}
}
}
break;
case paInt32:
{
int *out = (int *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = ((int) phase ) << 16;
if( data->numChannels == 2 )
{
*out++ = ((int) phase ) << 16;
}
}
}
break;
case paInt16:
{
short *out = (short *) outputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
phase += 0x123;
*out++ = phase;
if( data->numChannels == 2 )
{
*out++ = phase;
}
}
}
break;
default:
{
unsigned char *out = (unsigned char *) outputBuffer;
unsigned long numBytes = framesPerBuffer * data->numChannels * data->bytesPerSample;
for( i=0; i<numBytes; i++ )
{
*out++ = 0;
}
}
break;
}
data->sawPhase = phase;
}
/* Are we through yet? */
if( data->framesLeft > framesPerBuffer )
{
AddTraceMessage("QaCallback: running. framesLeft", data->framesLeft );
data->framesLeft -= framesPerBuffer;
return 0;
}
else
{
AddTraceMessage("QaCallback: DONE! framesLeft", data->framesLeft );
data->framesLeft = 0;
return 1;
}
}
