*/
+#include <assert.h>
#include <stdio.h>
#include <string.h> /* strlen() */
+
+#include <initguid.h> /* make sure ds guids get defined */
#include <windows.h>
#include <objbase.h>
/*
- We are only using DX3 in here, no need to polute the namespace - davidv
+ Use the earliest version of DX required, no need to polute the namespace
*/
+#ifdef PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE
+#define DIRECTSOUND_VERSION 0x0800
+#else
#define DIRECTSOUND_VERSION 0x0300
+#endif
#include <dsound.h>
#ifdef PAWIN_USE_WDMKS_DEVICE_INFO
#include <dsconf.h>
PaUtilBufferProcessor bufferProcessor;
/* DirectSound specific data. */
+#ifdef PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE
+ LPDIRECTSOUNDFULLDUPLEX8 pDirectSoundFullDuplex8;
+#endif
/* Output */
LPDIRECTSOUND pDirectSound;
UINT previousPlayCursor;
UINT outputUnderflowCount;
BOOL outputIsRunning;
- /* use double which lets us can play for several thousand years with enough precision */
- double dsw_framesWritten;
INT finalZeroBytesWritten; /* used to determine when we've flushed the whole buffer */
/* Input */
}
-static HRESULT InitInputBuffer( PaWinDsStream *stream, PaSampleFormat sampleFormat, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer, PaWinWaveFormatChannelMask channelMask )
+#ifdef PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE
+static HRESULT InitFullDuplexInputOutputBuffers( PaWinDsStream *stream,
+ PaWinDsDeviceInfo *inputDevice,
+ PaSampleFormat hostInputSampleFormat,
+ WORD inputChannelCount,
+ int bytesPerInputBuffer,
+ PaWinWaveFormatChannelMask inputChannelMask,
+ PaWinDsDeviceInfo *outputDevice,
+ PaSampleFormat hostOutputSampleFormat,
+ WORD outputChannelCount,
+ int bytesPerOutputBuffer,
+ PaWinWaveFormatChannelMask outputChannelMask,
+ unsigned long nFrameRate
+ )
+{
+ HRESULT hr;
+ DSCBUFFERDESC captureDesc;
+ PaWinWaveFormat captureWaveFormat;
+ DSBUFFERDESC secondaryRenderDesc;
+ PaWinWaveFormat renderWaveFormat;
+ LPDIRECTSOUNDBUFFER8 pRenderBuffer8;
+ LPDIRECTSOUNDCAPTUREBUFFER8 pCaptureBuffer8;
+
+ // capture buffer description
+
+ // only try wave format extensible. assume it's available on all ds 8 systems
+ PaWin_InitializeWaveFormatExtensible( &captureWaveFormat, inputChannelCount,
+ hostInputSampleFormat, PaWin_SampleFormatToLinearWaveFormatTag( hostInputSampleFormat ),
+ nFrameRate, inputChannelMask );
+
+ ZeroMemory(&captureDesc, sizeof(DSCBUFFERDESC));
+ captureDesc.dwSize = sizeof(DSCBUFFERDESC);
+ captureDesc.dwFlags = 0;
+ captureDesc.dwBufferBytes = bytesPerInputBuffer;
+ captureDesc.lpwfxFormat = (WAVEFORMATEX*)&captureWaveFormat;
+
+ // render buffer description
+
+ PaWin_InitializeWaveFormatExtensible( &renderWaveFormat, outputChannelCount,
+ hostOutputSampleFormat, PaWin_SampleFormatToLinearWaveFormatTag( hostOutputSampleFormat ),
+ nFrameRate, outputChannelMask );
+
+ ZeroMemory(&secondaryRenderDesc, sizeof(DSBUFFERDESC));
+ secondaryRenderDesc.dwSize = sizeof(DSBUFFERDESC);
+ secondaryRenderDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+ secondaryRenderDesc.dwBufferBytes = bytesPerOutputBuffer;
+ secondaryRenderDesc.lpwfxFormat = (WAVEFORMATEX*)&renderWaveFormat;
+
+ /* note that we don't create a primary buffer here at all */
+
+ hr = paWinDsDSoundEntryPoints.DirectSoundFullDuplexCreate8(
+ inputDevice->lpGUID, outputDevice->lpGUID,
+ &captureDesc, &secondaryRenderDesc,
+ GetDesktopWindow(), /* see InitOutputBuffer() for a discussion of whether this is a good idea */
+ DSSCL_EXCLUSIVE,
+ &stream->pDirectSoundFullDuplex8,
+ &pCaptureBuffer8,
+ &pRenderBuffer8,
+ NULL /* pUnkOuter must be NULL */
+ );
+
+ if( hr == DS_OK )
+ {
+ PA_DEBUG(("DirectSoundFullDuplexCreate succeeded!\n"));
+
+ /* retrieve the pre ds 8 buffer interfaces which are used by the rest of the code */
+
+ hr = IUnknown_QueryInterface( pCaptureBuffer8, &IID_IDirectSoundCaptureBuffer, &stream->pDirectSoundInputBuffer );
+
+ if( hr == DS_OK )
+ hr = IUnknown_QueryInterface( pRenderBuffer8, &IID_IDirectSoundBuffer, &stream->pDirectSoundOutputBuffer );
+
+ /* release the ds 8 interfaces, we don't need them */
+ IUnknown_Release( pCaptureBuffer8 );
+ IUnknown_Release( pRenderBuffer8 );
+
+ if( !stream->pDirectSoundInputBuffer || !stream->pDirectSoundOutputBuffer ){
+ /* couldn't get pre ds 8 interfaces for some reason. clean up. */
+ if( stream->pDirectSoundInputBuffer )
+ {
+ IUnknown_Release( stream->pDirectSoundInputBuffer );
+ stream->pDirectSoundInputBuffer = NULL;
+ }
+
+ if( stream->pDirectSoundOutputBuffer )
+ {
+ IUnknown_Release( stream->pDirectSoundOutputBuffer );
+ stream->pDirectSoundOutputBuffer = NULL;
+ }
+
+ IUnknown_Release( stream->pDirectSoundFullDuplex8 );
+ stream->pDirectSoundFullDuplex8 = NULL;
+ }
+ }
+ else
+ {
+ PA_DEBUG(("DirectSoundFullDuplexCreate failed. hr=%d\n", hr));
+ }
+
+ return hr;
+}
+#endif /* PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE */
+
+
+static HRESULT InitInputBuffer( PaWinDsStream *stream, PaWinDsDeviceInfo *device, PaSampleFormat sampleFormat, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer, PaWinWaveFormatChannelMask channelMask )
{
DSCBUFFERDESC captureDesc;
PaWinWaveFormat waveFormat;
HRESULT result;
- stream->bytesPerInputFrame = nChannels * Pa_GetSampleSize(sampleFormat);
+ if( (result = paWinDsDSoundEntryPoints.DirectSoundCaptureCreate(
+ device->lpGUID, &stream->pDirectSoundCapture, NULL) ) != DS_OK ){
+ ERR_RPT(("PortAudio: DirectSoundCaptureCreate() failed!\n"));
+ return result;
+ }
- stream->inputSize = bytesPerBuffer;
- // ----------------------------------------------------------------------
// Setup the secondary buffer description
ZeroMemory(&captureDesc, sizeof(DSCBUFFERDESC));
captureDesc.dwSize = sizeof(DSCBUFFERDESC);
- captureDesc.dwFlags = 0;
+ captureDesc.dwFlags = 0;
captureDesc.dwBufferBytes = bytesPerBuffer;
captureDesc.lpwfxFormat = (WAVEFORMATEX*)&waveFormat;
}
-static HRESULT InitOutputBuffer( PaWinDsStream *stream, PaSampleFormat sampleFormat, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer, PaWinWaveFormatChannelMask channelMask )
+static HRESULT InitOutputBuffer( PaWinDsStream *stream, PaWinDsDeviceInfo *device, PaSampleFormat sampleFormat, unsigned long nFrameRate, WORD nChannels, int bytesPerBuffer, PaWinWaveFormatChannelMask channelMask )
{
HRESULT result;
HWND hWnd;
PaWinWaveFormat waveFormat;
DSBUFFERDESC primaryDesc;
DSBUFFERDESC secondaryDesc;
- LARGE_INTEGER counterFrequency;
- int bytesPerSample = Pa_GetSampleSize(sampleFormat);
-
- stream->outputBufferSizeBytes = bytesPerBuffer;
- stream->outputIsRunning = FALSE;
- stream->outputUnderflowCount = 0;
- stream->dsw_framesWritten = 0;
- stream->bytesPerOutputFrame = nChannels * bytesPerSample;
-
- /* perfCounterTicksPerBuffer is used by QueryOutputSpace for overflow detection */
- if( QueryPerformanceFrequency( &counterFrequency ) )
- {
- int framesInBuffer = bytesPerBuffer / (nChannels * bytesPerSample);
- stream->perfCounterTicksPerBuffer.QuadPart = (counterFrequency.QuadPart * framesInBuffer) / nFrameRate;
- }
- else
- {
- stream->perfCounterTicksPerBuffer.QuadPart = 0;
+
+ if( (hr = paWinDsDSoundEntryPoints.DirectSoundCreate(
+ device->lpGUID, &stream->pDirectSound, NULL )) != DS_OK ){
+ ERR_RPT(("PortAudio: DirectSoundCreate() failed!\n"));
+ return hr;
}
-
// 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.)
return hr;
}
- stream->pDirectSoundPrimaryBuffer = NULL;
- stream->pDirectSoundOutputBuffer = NULL;
-
// -----------------------------------------------------------------------
// 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 secondary buffer description
ZeroMemory(&secondaryDesc, sizeof(DSBUFFERDESC));
secondaryDesc.dwSize = sizeof(DSBUFFERDESC);
- secondaryDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+ secondaryDesc.dwFlags = DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2;
secondaryDesc.dwBufferBytes = bytesPerBuffer;
secondaryDesc.lpwfxFormat = (WAVEFORMATEX*)&waveFormat;
// Create the secondary buffer
/* DirectSound specific initialization */
{
HRESULT hr;
- int bytesPerDirectSoundBuffer;
+ int bytesPerDirectSoundInputBuffer, bytesPerDirectSoundOutputBuffer;
int userLatencyFrames;
int minLatencyFrames;
-
+ unsigned long integerSampleRate = (unsigned long) (sampleRate + 0.5);
+
stream->timerID = 0;
stream->processingCompleted = CreateEvent( NULL, /* bManualReset = */ TRUE, /* bInitialState = */ FALSE, NULL );
PRINT(("PortAudio on DirectSound - Latency = %d frames, %d msec\n", stream->framesPerDSBuffer, msecLatency ));
}
+ /* set up i/o parameters */
/* ------------------ OUTPUT */
if( outputParameters )
{
+ LARGE_INTEGER counterFrequency;
+
/*
PaDeviceInfo *deviceInfo = hostApi->deviceInfos[ outputParameters->device ];
DBUG(("PaHost_OpenStream: deviceID = 0x%x\n", outputParameters->device));
*/
int bytesPerSample = Pa_GetSampleSize(hostOutputSampleFormat);
- bytesPerDirectSoundBuffer = stream->framesPerDSBuffer * outputParameters->channelCount * bytesPerSample;
- if( bytesPerDirectSoundBuffer < DSBSIZE_MIN )
+ bytesPerDirectSoundOutputBuffer = stream->framesPerDSBuffer * outputParameters->channelCount * bytesPerSample;
+ if( bytesPerDirectSoundOutputBuffer < DSBSIZE_MIN )
{
result = paBufferTooSmall;
goto error;
}
- else if( bytesPerDirectSoundBuffer > DSBSIZE_MAX )
+ else if( bytesPerDirectSoundOutputBuffer > DSBSIZE_MAX )
{
result = paBufferTooBig;
goto error;
}
- hr = paWinDsDSoundEntryPoints.DirectSoundCreate(
- ((PaWinDsDeviceInfo*)hostApi->deviceInfos[outputParameters->device])->lpGUID,
- &stream->pDirectSound, NULL );
+ /* Calculate value used in latency calculation to avoid real-time divides. */
+ stream->secondsPerHostByte = 1.0 /
+ (stream->bufferProcessor.bytesPerHostOutputSample *
+ outputChannelCount * sampleRate);
- if( hr != DS_OK )
+ stream->outputBufferSizeBytes = bytesPerDirectSoundOutputBuffer;
+ stream->outputIsRunning = FALSE;
+ stream->outputUnderflowCount = 0;
+ stream->bytesPerOutputFrame = outputParameters->channelCount * bytesPerSample;
+
+ /* perfCounterTicksPerBuffer is used by QueryOutputSpace for overflow detection */
+ if( QueryPerformanceFrequency( &counterFrequency ) )
{
- ERR_RPT(("PortAudio: DirectSoundCreate() failed!\n"));
- result = paUnanticipatedHostError;
- PA_DS_SET_LAST_DIRECTSOUND_ERROR( hr );
- goto error;
+ stream->perfCounterTicksPerBuffer.QuadPart = (counterFrequency.QuadPart * stream->framesPerDSBuffer) / integerSampleRate;
}
- hr = InitOutputBuffer( stream,
- hostOutputSampleFormat,
- (unsigned long) (sampleRate + 0.5),
- (WORD)outputParameters->channelCount, bytesPerDirectSoundBuffer,
- outputChannelMask );
- DBUG(("InitOutputBuffer() returns %x\n", hr));
- if( hr != DS_OK )
+ else
{
- result = paUnanticipatedHostError;
- PA_DS_SET_LAST_DIRECTSOUND_ERROR( hr );
- goto error;
+ stream->perfCounterTicksPerBuffer.QuadPart = 0;
}
- /* Calculate value used in latency calculation to avoid real-time divides. */
- stream->secondsPerHostByte = 1.0 /
- (stream->bufferProcessor.bytesPerHostOutputSample *
- outputChannelCount * sampleRate);
}
/* ------------------ INPUT */
*/
int bytesPerSample = Pa_GetSampleSize(hostInputSampleFormat);
- bytesPerDirectSoundBuffer = stream->framesPerDSBuffer * inputParameters->channelCount * bytesPerSample;
- if( bytesPerDirectSoundBuffer < DSBSIZE_MIN )
+ bytesPerDirectSoundInputBuffer = stream->framesPerDSBuffer * inputParameters->channelCount * bytesPerSample;
+ if( bytesPerDirectSoundInputBuffer < DSBSIZE_MIN )
{
result = paBufferTooSmall;
goto error;
}
- else if( bytesPerDirectSoundBuffer > DSBSIZE_MAX )
+ else if( bytesPerDirectSoundInputBuffer > DSBSIZE_MAX )
{
result = paBufferTooBig;
goto error;
}
- hr = paWinDsDSoundEntryPoints.DirectSoundCaptureCreate(
- ((PaWinDsDeviceInfo*)hostApi->deviceInfos[inputParameters->device])->lpGUID,
- &stream->pDirectSoundCapture, NULL );
+ stream->bytesPerInputFrame = inputParameters->channelCount * bytesPerSample;
+
+ stream->inputSize = bytesPerDirectSoundInputBuffer;
+ }
+
+ /* open/create the DirectSound buffers */
+
+ /* interface ptrs should be zeroed when stream is zeroed. */
+ assert( stream->pDirectSoundCapture == NULL );
+ assert( stream->pDirectSoundInputBuffer == NULL );
+ assert( stream->pDirectSound == NULL );
+ assert( stream->pDirectSoundPrimaryBuffer == NULL );
+ assert( stream->pDirectSoundOutputBuffer == NULL );
+
+
+ if( inputParameters && outputParameters )
+ {
+#ifdef PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE
+ /* try to use the full-duplex DX8 API to create the buffers.
+ if that fails we fall back to the half-duplex API below */
+
+ hr = InitFullDuplexInputOutputBuffers( stream,
+ (PaWinDsDeviceInfo*)hostApi->deviceInfos[inputParameters->device],
+ hostInputSampleFormat,
+ (WORD)inputParameters->channelCount, bytesPerDirectSoundInputBuffer,
+ inputChannelMask,
+ (PaWinDsDeviceInfo*)hostApi->deviceInfos[outputParameters->device],
+ hostOutputSampleFormat,
+ (WORD)outputParameters->channelCount, bytesPerDirectSoundOutputBuffer,
+ outputChannelMask,
+ integerSampleRate
+ );
+ DBUG(("InitFullDuplexInputOutputBuffers() returns %x\n", hr));
+ /* ignore any error returned by InitFullDuplexInputOutputBuffers.
+ we retry opening the buffers below */
+#endif /* PAWIN_USE_DIRECTSOUNDFULLDUPLEXCREATE */
+ }
+
+ /* create half duplex buffers. also used for full-duplex streams which didn't
+ succeed when using the full duplex API. that could happen because
+ DX8 or greater isnt installed, the i/o devices aren't the same
+ physical device. etc.
+ */
+
+ if( outputParameters && !stream->pDirectSoundOutputBuffer )
+ {
+ hr = InitOutputBuffer( stream,
+ (PaWinDsDeviceInfo*)hostApi->deviceInfos[outputParameters->device],
+ hostOutputSampleFormat,
+ integerSampleRate,
+ (WORD)outputParameters->channelCount, bytesPerDirectSoundOutputBuffer,
+ outputChannelMask );
+ DBUG(("InitOutputBuffer() returns %x\n", hr));
if( hr != DS_OK )
{
- ERR_RPT(("PortAudio: DirectSoundCaptureCreate() failed!\n"));
result = paUnanticipatedHostError;
PA_DS_SET_LAST_DIRECTSOUND_ERROR( hr );
goto error;
}
+ }
+
+ if( inputParameters && !stream->pDirectSoundInputBuffer )
+ {
hr = InitInputBuffer( stream,
+ (PaWinDsDeviceInfo*)hostApi->deviceInfos[inputParameters->device],
hostInputSampleFormat,
- (unsigned long) (sampleRate + 0.5),
- (WORD)inputParameters->channelCount, bytesPerDirectSoundBuffer,
+ integerSampleRate,
+ (WORD)inputParameters->channelCount, bytesPerDirectSoundInputBuffer,
inputChannelMask );
DBUG(("InitInputBuffer() returns %x\n", hr));
if( hr != DS_OK )
goto error;
}
}
-
}
*s = (PaStream*)stream;
stream->pDirectSound = NULL;
}
+ if( stream->pDirectSoundFullDuplex8 )
+ {
+ IDirectSoundFullDuplex_Release( stream->pDirectSoundFullDuplex8 );
+ stream->pDirectSoundFullDuplex8 = NULL;
+ }
+
if( bufferProcessorIsInitialized )
PaUtil_TerminateBufferProcessor( &stream->bufferProcessor );
}
// FIXME: what happens if IDirectSoundCaptureBuffer_GetCurrentPosition fails?
- framesToXfer = numInFramesReady = bytesFilled / stream->bytesPerInputFrame;
- outputLatency = ((double)bytesFilled) * stream->secondsPerHostByte;
+ framesToXfer = numInFramesReady = bytesFilled / stream->bytesPerInputFrame;
+ outputLatency = ((double)bytesFilled) * stream->secondsPerHostByte; // FIXME: this doesn't look right. we're calculating output latency in input branch. also secondsPerHostByte is only initialized for the output stream
/** @todo Check for overflow */
}
/* The outputBufferDacTime parameter should indicates the time at which
the first sample of the output buffer is heard at the DACs. */
timeInfo.currentTime = PaUtil_GetTime();
- timeInfo.outputBufferDacTime = timeInfo.currentTime + outputLatency;
+ timeInfo.outputBufferDacTime = timeInfo.currentTime + outputLatency; // FIXME: QueryOutputSpace gets the playback position, we could use that (?)
PaUtil_BeginBufferProcessing( &stream->bufferProcessor, &timeInfo, stream->callbackFlags );
bytesProcessed = numFrames * stream->bytesPerOutputFrame;
stream->outputBufferWriteOffsetBytes = (stream->outputBufferWriteOffsetBytes + bytesProcessed) % stream->outputBufferSizeBytes;
IDirectSoundBuffer_Unlock( stream->pDirectSoundOutputBuffer, lpOutBuf1, dwOutSize1, lpOutBuf2, dwOutSize2);
- stream->dsw_framesWritten += numFrames;
}
error1:
// Update our buffer offset and unlock sound buffer
stream->outputBufferWriteOffsetBytes = (stream->outputBufferWriteOffsetBytes + dwsize1 + dwsize2) % stream->outputBufferSizeBytes;
IDirectSoundBuffer_Unlock( stream->pDirectSoundOutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2);
- stream->dsw_framesWritten += bytesEmpty / stream->bytesPerOutputFrame;
stream->finalZeroBytesWritten += dwsize1 + dwsize2;
}
stream->pDirectSound = NULL;
}
+ if( stream->pDirectSoundFullDuplex8 )
+ {
+ IDirectSoundFullDuplex_Release( stream->pDirectSoundFullDuplex8 );
+ stream->pDirectSoundFullDuplex8 = NULL;
+ }
+
PaUtil_TerminateBufferProcessor( &stream->bufferProcessor );
PaUtil_TerminateStreamRepresentation( &stream->streamRepresentation );
PaUtil_FreeMemory( stream );
&stream->previousPlayCursor, &stream->outputBufferWriteOffsetBytes )) != DS_OK)
return hr;
- stream->dsw_framesWritten = stream->outputBufferWriteOffsetBytes / stream->bytesPerOutputFrame;
/* printf("DSW_InitOutputBuffer: playCursor = %d, writeCursor = %d\n", playCursor, dsw->dsw_WriteOffset ); */
return DS_OK;
// FIXME: what happens if IDirectSoundBuffer_Stop returns an error?
hr = IDirectSoundBuffer_Stop( stream->pDirectSoundOutputBuffer );
- IDirectSoundBuffer_Stop( stream->pDirectSoundPrimaryBuffer ); /* FIXME we never started the primary buffer so I'm not sure we need to stop it */
+ if( stream->pDirectSoundPrimaryBuffer )
+ IDirectSoundBuffer_Stop( stream->pDirectSoundPrimaryBuffer ); /* FIXME we never started the primary buffer so I'm not sure we need to stop it */
}
}
projects / portaudio / commitdiff