diff options
Diffstat (limited to 'pd/portaudio_v18/pa_win_ds')
| -rw-r--r-- | pd/portaudio_v18/pa_win_ds/dsound_wrapper.c | 466 | ||||
| -rw-r--r-- | pd/portaudio_v18/pa_win_ds/dsound_wrapper.h | 106 | ||||
| -rw-r--r-- | pd/portaudio_v18/pa_win_ds/pa_dsound.c | 1042 | ||||
| -rw-r--r-- | pd/portaudio_v18/pa_win_ds/portaudio.def | 28 |
4 files changed, 1642 insertions, 0 deletions
diff --git a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c new file mode 100644 index 00000000..527f0eb3 --- /dev/null +++ b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c @@ -0,0 +1,466 @@ +/* + * $Id: dsound_wrapper.c,v 1.1.1.1 2002/01/22 00:52:45 phil Exp $ + * Simplified DirectSound interface. + * + * Author: Phil Burk & Robert Marsanyi + * + * PortAudio Portable Real-Time Audio Library + * For more information see: http://www.softsynth.com/portaudio/ + * DirectSound Implementation + * Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * Any person wishing to distribute modifications to the Software is + * requested to send the modifications to the original developer so that + * they can be incorporated into the canonical version. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + * + */ +#include <stdio.h> +#include <stdlib.h> +#include <math.h> +#define INITGUID // Needed to build IID_IDirectSoundNotify. See objbase.h for info. +#include <objbase.h> +#include <unknwn.h> +#include "dsound_wrapper.h" +#include "pa_trace.h" + +/************************************************************************************/ +void DSW_Term( DSoundWrapper *dsw ) +{ + // Cleanup the sound buffers + if (dsw->dsw_OutputBuffer) + { + IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer ); + IDirectSoundBuffer_Release( dsw->dsw_OutputBuffer ); + dsw->dsw_OutputBuffer = NULL; + } +#if SUPPORT_AUDIO_CAPTURE + if (dsw->dsw_InputBuffer) + { + IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer ); + IDirectSoundCaptureBuffer_Release( dsw->dsw_InputBuffer ); + dsw->dsw_InputBuffer = NULL; + } + if (dsw->dsw_pDirectSoundCapture) + { + IDirectSoundCapture_Release( dsw->dsw_pDirectSoundCapture ); + dsw->dsw_pDirectSoundCapture = NULL; + } +#endif /* SUPPORT_AUDIO_CAPTURE */ + if (dsw->dsw_pDirectSound) + { + IDirectSound_Release( dsw->dsw_pDirectSound ); + dsw->dsw_pDirectSound = NULL; + } +} +/************************************************************************************/ +HRESULT DSW_Init( DSoundWrapper *dsw ) +{ + memset( dsw, 0, sizeof(DSoundWrapper) ); + return 0; +} +/************************************************************************************/ +HRESULT DSW_InitOutputDevice( DSoundWrapper *dsw, LPGUID lpGUID ) +{ + // Create the DS object + HRESULT hr = DirectSoundCreate( lpGUID, &dsw->dsw_pDirectSound, NULL ); + if( hr != DS_OK ) return hr; + return hr; +} + +/************************************************************************************/ +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_StartOutput( DSoundWrapper *dsw ) +{ + HRESULT hr; + QueryPerformanceCounter( &dsw->dsw_LastPlayTime ); + dsw->dsw_LastPlayCursor = 0; + dsw->dsw_FramesPlayed = 0; + hr = IDirectSoundBuffer_SetCurrentPosition( dsw->dsw_OutputBuffer, 0 ); + if( hr != DS_OK ) + { + return hr; + } + // Start the buffer playback in a loop. + if( dsw->dsw_OutputBuffer != NULL ) + { + hr = IDirectSoundBuffer_Play( dsw->dsw_OutputBuffer, 0, 0, DSBPLAY_LOOPING ); + if( hr != DS_OK ) + { + return hr; + } + dsw->dsw_OutputRunning = TRUE; + } + + return 0; +} +/************************************************************************************/ +HRESULT DSW_StopOutput( DSoundWrapper *dsw ) +{ + // Stop the buffer playback + if( dsw->dsw_OutputBuffer != NULL ) + { + dsw->dsw_OutputRunning = FALSE; + return IDirectSoundBuffer_Stop( dsw->dsw_OutputBuffer ); + } + else return 0; +} + +/************************************************************************************/ +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; +} + +/************************************************************************************/ +HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw ) +{ + HRESULT hr; + LPBYTE lpbuf1 = NULL; + LPBYTE lpbuf2 = NULL; + DWORD dwsize1 = 0; + DWORD dwsize2 = 0; + long bytesEmpty; + hr = DSW_QueryOutputSpace( dsw, &bytesEmpty ); // updates dsw_FramesPlayed + if (hr != DS_OK) return hr; + if( bytesEmpty == 0 ) return DS_OK; + // Lock free space in the DS + hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, bytesEmpty, (void **) &lpbuf1, &dwsize1, + (void **) &lpbuf2, &dwsize2, 0); + if (hr == DS_OK) + { + // Copy the buffer into the DS + ZeroMemory(lpbuf1, dwsize1); + if(lpbuf2 != NULL) + { + ZeroMemory(lpbuf2, dwsize2); + } + // Update our buffer offset and unlock sound buffer + dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize; + IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); + dsw->dsw_FramesWritten += bytesEmpty / dsw->dsw_BytesPerFrame; + } + return hr; +} + +/************************************************************************************/ +HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes ) +{ + HRESULT hr; + LPBYTE lpbuf1 = NULL; + LPBYTE lpbuf2 = NULL; + DWORD dwsize1 = 0; + DWORD dwsize2 = 0; + // Lock free space in the DS + hr = IDirectSoundBuffer_Lock( dsw->dsw_OutputBuffer, dsw->dsw_WriteOffset, numBytes, (void **) &lpbuf1, &dwsize1, + (void **) &lpbuf2, &dwsize2, 0); + if (hr == DS_OK) + { + // Copy the buffer into the DS + CopyMemory(lpbuf1, buf, dwsize1); + if(lpbuf2 != NULL) + { + CopyMemory(lpbuf2, buf+dwsize1, dwsize2); + } + // Update our buffer offset and unlock sound buffer + dsw->dsw_WriteOffset = (dsw->dsw_WriteOffset + dwsize1 + dwsize2) % dsw->dsw_OutputSize; + IDirectSoundBuffer_Unlock( dsw->dsw_OutputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); + dsw->dsw_FramesWritten += numBytes / dsw->dsw_BytesPerFrame; + } + return hr; +} + +/************************************************************************************/ +DWORD DSW_GetOutputStatus( DSoundWrapper *dsw ) +{ + DWORD status; + if (IDirectSoundBuffer_GetStatus( dsw->dsw_OutputBuffer, &status ) != DS_OK) + return( DSERR_INVALIDPARAM ); + else + return( status ); +} + +#if SUPPORT_AUDIO_CAPTURE +/* These routines are used to support audio input. + * Do NOT compile these calls when using NT4 because it does + * not support the entry points. + */ +/************************************************************************************/ +HRESULT DSW_InitInputDevice( DSoundWrapper *dsw, LPGUID lpGUID ) +{ + HRESULT hr = DirectSoundCaptureCreate( lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); + if( hr != DS_OK ) return hr; + return hr; +} +/************************************************************************************/ +HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, int nChannels, int bytesPerBuffer ) +{ + DSCBUFFERDESC captureDesc; + WAVEFORMATEX wfFormat; + HRESULT 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. */ + dsw->dsw_InputSize = bytesPerBuffer; + // ---------------------------------------------------------------------- + // Setup the secondary buffer description + ZeroMemory(&captureDesc, sizeof(DSCBUFFERDESC)); + captureDesc.dwSize = sizeof(DSCBUFFERDESC); + captureDesc.dwFlags = 0; + captureDesc.dwBufferBytes = bytesPerBuffer; + captureDesc.lpwfxFormat = &wfFormat; + // Create the capture buffer + if ((result = IDirectSoundCapture_CreateCaptureBuffer( dsw->dsw_pDirectSoundCapture, + &captureDesc, &dsw->dsw_InputBuffer, NULL)) != DS_OK) return result; + dsw->dsw_ReadOffset = 0; // reset last read position to start of buffer + return DS_OK; +} + +/************************************************************************************/ +HRESULT DSW_StartInput( DSoundWrapper *dsw ) +{ + // Start the buffer playback + if( dsw->dsw_InputBuffer != NULL ) + { + return IDirectSoundCaptureBuffer_Start( dsw->dsw_InputBuffer, DSCBSTART_LOOPING ); + } + else return 0; +} + +/************************************************************************************/ +HRESULT DSW_StopInput( DSoundWrapper *dsw ) +{ + // Stop the buffer playback + if( dsw->dsw_InputBuffer != NULL ) + { + return IDirectSoundCaptureBuffer_Stop( dsw->dsw_InputBuffer ); + } + else return 0; +} + +/************************************************************************************/ +HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled ) +{ + HRESULT hr; + DWORD capturePos; + DWORD readPos; + long filled; + // Query to see how much data is in buffer. + // We don't need the capture position but sometimes DirectSound doesn't handle NULLS correctly + // so let's pass a pointer just to be safe. + hr = IDirectSoundCaptureBuffer_GetCurrentPosition( dsw->dsw_InputBuffer, &capturePos, &readPos ); + if( hr != DS_OK ) + { + return hr; + } + filled = readPos - dsw->dsw_ReadOffset; + if( filled < 0 ) filled += dsw->dsw_InputSize; // unwrap offset + *bytesFilled = filled; + return hr; +} + +/************************************************************************************/ +HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes ) +{ + HRESULT hr; + LPBYTE lpbuf1 = NULL; + LPBYTE lpbuf2 = NULL; + DWORD dwsize1 = 0; + DWORD dwsize2 = 0; + // Lock free space in the DS + hr = IDirectSoundCaptureBuffer_Lock ( dsw->dsw_InputBuffer, dsw->dsw_ReadOffset, numBytes, (void **) &lpbuf1, &dwsize1, + (void **) &lpbuf2, &dwsize2, 0); + if (hr == DS_OK) + { + // Copy from DS to the buffer + CopyMemory( buf, lpbuf1, dwsize1); + if(lpbuf2 != NULL) + { + CopyMemory( buf+dwsize1, lpbuf2, dwsize2); + } + // Update our buffer offset and unlock sound buffer + dsw->dsw_ReadOffset = (dsw->dsw_ReadOffset + dwsize1 + dwsize2) % dsw->dsw_InputSize; + IDirectSoundCaptureBuffer_Unlock ( dsw->dsw_InputBuffer, lpbuf1, dwsize1, lpbuf2, dwsize2); + } + return hr; +} + +#endif /* SUPPORT_AUDIO_CAPTURE */ diff --git a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h new file mode 100644 index 00000000..b04b0020 --- /dev/null +++ b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h @@ -0,0 +1,106 @@ +#ifndef __DSOUND_WRAPPER_H +#define __DSOUND_WRAPPER_H +/* + * $Id: dsound_wrapper.h,v 1.1.1.1 2002/01/22 00:52:45 phil Exp $ + * Simplified DirectSound interface. + * + * Author: Phil Burk & Robert Marsanyi + * + * For PortAudio Portable Real-Time Audio Library + * For more information see: http://www.softsynth.com/portaudio/ + * DirectSound Implementation + * Copyright (c) 1999-2000 Phil Burk & Robert Marsanyi + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * Any person wishing to distribute modifications to the Software is + * requested to send the modifications to the original developer so that + * they can be incorporated into the canonical version. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + * + */ +#include <DSound.h> +#if !defined(BOOL) +#define BOOL short +#endif +#ifndef SUPPORT_AUDIO_CAPTURE +#define SUPPORT_AUDIO_CAPTURE (1) +#endif + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#define DSW_NUM_POSITIONS (4) +#define DSW_NUM_EVENTS (5) +#define DSW_TERMINATION_EVENT (DSW_NUM_POSITIONS) + +typedef struct +{ + /* Output */ + LPDIRECTSOUND dsw_pDirectSound; + LPDIRECTSOUNDBUFFER dsw_OutputBuffer; + DWORD dsw_WriteOffset; /* last write position */ + INT dsw_OutputSize; + INT dsw_BytesPerFrame; + /* Try to detect play buffer underflows. */ + LARGE_INTEGER dsw_CounterTicksPerBuffer; /* counter ticks it should take to play a full buffer */ + LARGE_INTEGER dsw_LastPlayTime; + UINT dsw_LastPlayCursor; + UINT dsw_OutputUnderflows; + BOOL dsw_OutputRunning; + /* use double which lets us can play for several thousand years with enough precision */ + double dsw_FramesWritten; + double dsw_FramesPlayed; +#if SUPPORT_AUDIO_CAPTURE + /* Input */ + LPDIRECTSOUNDCAPTURE dsw_pDirectSoundCapture; + LPDIRECTSOUNDCAPTUREBUFFER dsw_InputBuffer; + UINT dsw_ReadOffset; /* last read position */ + UINT dsw_InputSize; +#endif /* SUPPORT_AUDIO_CAPTURE */ + +} +DSoundWrapper; +HRESULT DSW_Init( DSoundWrapper *dsw ); +void DSW_Term( DSoundWrapper *dsw ); +HRESULT DSW_InitOutputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, + int nChannels, int bufSize ); +HRESULT DSW_StartOutput( DSoundWrapper *dsw ); +HRESULT DSW_StopOutput( DSoundWrapper *dsw ); +DWORD DSW_GetOutputStatus( DSoundWrapper *dsw ); +HRESULT DSW_WriteBlock( DSoundWrapper *dsw, char *buf, long numBytes ); +HRESULT DSW_ZeroEmptySpace( DSoundWrapper *dsw ); +HRESULT DSW_QueryOutputSpace( DSoundWrapper *dsw, long *bytesEmpty ); +HRESULT DSW_Enumerate( DSoundWrapper *dsw ); + +#if SUPPORT_AUDIO_CAPTURE +HRESULT DSW_InitInputBuffer( DSoundWrapper *dsw, unsigned long nFrameRate, + int nChannels, int bufSize ); +HRESULT DSW_StartInput( DSoundWrapper *dsw ); +HRESULT DSW_StopInput( DSoundWrapper *dsw ); +HRESULT DSW_ReadBlock( DSoundWrapper *dsw, char *buf, long numBytes ); +HRESULT DSW_QueryInputFilled( DSoundWrapper *dsw, long *bytesFilled ); +#endif /* SUPPORT_AUDIO_CAPTURE */ + +#ifdef __cplusplus +} +#endif /* __cplusplus */ +#endif /* __DSOUND_WRAPPER_H */ diff --git a/pd/portaudio_v18/pa_win_ds/pa_dsound.c b/pd/portaudio_v18/pa_win_ds/pa_dsound.c new file mode 100644 index 00000000..52532625 --- /dev/null +++ b/pd/portaudio_v18/pa_win_ds/pa_dsound.c @@ -0,0 +1,1042 @@ +/* + * $Id: pa_dsound.c,v 1.2.4.1 2002/11/19 20:46:06 philburk Exp $ + * PortAudio Portable Real-Time Audio Library + * Latest Version at: http://www.softsynth.com/portaudio/ + * DirectSound Implementation + * + * Copyright (c) 1999-2000 Phil Burk + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * Any person wishing to distribute modifications to the Software is + * requested to send the modifications to the original developer so that + * they can be incorporated into the canonical version. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + * + */ +/* Modifications + * 7/19/01 Mike Berry - casts for compiling with __MWERKS__ CodeWarrior + * 9/27/01 Phil Burk - use number of frames instead of real-time for CPULoad calculation. + * 4/19/02 Phil Burk - Check for Win XP for system latency calculation. + */ +/* Compiler flags: + SUPPORT_AUDIO_CAPTURE - define this flag if you want to SUPPORT_AUDIO_CAPTURE + */ + +#include <stdio.h> +#include <stdlib.h> +#ifndef __MWERKS__ +#include <malloc.h> +#include <memory.h> +#endif //__MWERKS__ +#include <math.h> +#include "portaudio.h" +#include "pa_host.h" +#include "pa_trace.h" +#include "dsound_wrapper.h" + +#define PRINT(x) { printf x; fflush(stdout); } +#define ERR_RPT(x) PRINT(x) +#define DBUG(x) /* PRINT(x) */ +#define DBUGX(x) /* PRINT(x) */ + +#define PA_USE_HIGH_LATENCY (0) +#if PA_USE_HIGH_LATENCY +#define PA_WIN_9X_LATENCY (500) +#define PA_WIN_NT_LATENCY (600) +#else +#define PA_WIN_9X_LATENCY (140) +#define PA_WIN_NT_LATENCY (280) +#endif + +#define PA_WIN_WDM_LATENCY (120) + +/* Trigger an underflow for testing purposes. Should normally be (0). */ +#define PA_SIMULATE_UNDERFLOW (0) +#if PA_SIMULATE_UNDERFLOW +static gUnderCallbackCounter = 0; +#define UNDER_START_GAP (10) +#define UNDER_STOP_GAP (UNDER_START_GAP + 4) +#endif + +/************************************************* Definitions ********/ +typedef struct internalPortAudioStream internalPortAudioStream; +typedef struct internalPortAudioDevice +{ + GUID pad_GUID; + GUID *pad_lpGUID; + double pad_SampleRates[10]; /* for pointing to from pad_Info FIXME?!*/ + PaDeviceInfo pad_Info; +} +internalPortAudioDevice; + +/* Define structure to contain all DirectSound and Windows specific data. */ +typedef struct PaHostSoundControl +{ + DSoundWrapper pahsc_DSoundWrapper; + MMRESULT pahsc_TimerID; + BOOL pahsc_IfInsideCallback; /* Test for reentrancy. */ + short *pahsc_NativeBuffer; + unsigned int pahsc_BytesPerBuffer; /* native buffer size in bytes */ + double pahsc_ValidFramesWritten; + int pahsc_FramesPerDSBuffer; + /* For measuring CPU utilization. */ + LARGE_INTEGER pahsc_EntryCount; + double pahsc_InverseTicksPerUserBuffer; +} +PaHostSoundControl; + +/************************************************* Shared Data ********/ +/* FIXME - put Mutex around this shared data. */ +static int sNumDevices = 0; +static int sDeviceIndex = 0; +static internalPortAudioDevice *sDevices = NULL; +static int sDefaultInputDeviceID = paNoDevice; +static int sDefaultOutputDeviceID = paNoDevice; +static int sEnumerationError; +static int sPaHostError = 0; +/************************************************* Prototypes **********/ +static internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ); +static BOOL CALLBACK Pa_EnumProc(LPGUID lpGUID, + LPCTSTR lpszDesc, + LPCTSTR lpszDrvName, + LPVOID lpContext ); +static BOOL CALLBACK Pa_CountDevProc(LPGUID lpGUID, + LPCTSTR lpszDesc, + LPCTSTR lpszDrvName, + LPVOID lpContext ); +static Pa_QueryDevices( void ); +static void CALLBACK Pa_TimerCallback(UINT uID, UINT uMsg, + DWORD dwUser, DWORD dw1, DWORD dw2); + +/********************************* BEGIN CPU UTILIZATION MEASUREMENT ****/ +static void Pa_StartUsageCalculation( internalPortAudioStream *past ) +{ + PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return; + /* Query system timer for usage analysis and to prevent overuse of CPU. */ + QueryPerformanceCounter( &pahsc->pahsc_EntryCount ); +} + +static void Pa_EndUsageCalculation( internalPortAudioStream *past ) +{ + LARGE_INTEGER CurrentCount = { 0, 0 }; + PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return; + /* + ** Measure CPU utilization during this callback. Note that this calculation + ** assumes that we had the processor the whole time. + */ +#define LOWPASS_COEFFICIENT_0 (0.9) +#define LOWPASS_COEFFICIENT_1 (0.99999 - LOWPASS_COEFFICIENT_0) + if( QueryPerformanceCounter( &CurrentCount ) ) + { + LONGLONG InsideCount = CurrentCount.QuadPart - pahsc->pahsc_EntryCount.QuadPart; + double newUsage = InsideCount * pahsc->pahsc_InverseTicksPerUserBuffer; + past->past_Usage = (LOWPASS_COEFFICIENT_0 * past->past_Usage) + + (LOWPASS_COEFFICIENT_1 * newUsage); + } +} + +/****************************************** END CPU UTILIZATION *******/ +static PaError Pa_QueryDevices( void ) +{ + int numBytes; + sDefaultInputDeviceID = paNoDevice; + sDefaultOutputDeviceID = paNoDevice; + /* Enumerate once just to count devices. */ + sNumDevices = 0; // for default device + DirectSoundEnumerate( (LPDSENUMCALLBACK)Pa_CountDevProc, NULL ); +#if SUPPORT_AUDIO_CAPTURE + DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK)Pa_CountDevProc, NULL ); +#endif /* SUPPORT_AUDIO_CAPTURE */ + /* Allocate structures to hold device info. */ + numBytes = sNumDevices * sizeof(internalPortAudioDevice); + sDevices = (internalPortAudioDevice *)PaHost_AllocateFastMemory( numBytes ); /* MEM */ + if( sDevices == NULL ) return paInsufficientMemory; + /* Enumerate again to fill in structures. */ + sDeviceIndex = 0; + sEnumerationError = 0; + DirectSoundEnumerate( (LPDSENUMCALLBACK)Pa_EnumProc, (void *)0 ); +#if SUPPORT_AUDIO_CAPTURE + if( sEnumerationError != paNoError ) return sEnumerationError; + sEnumerationError = 0; + DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK)Pa_EnumProc, (void *)1 ); +#endif /* SUPPORT_AUDIO_CAPTURE */ + return sEnumerationError; +} +/************************************************************************************/ +long Pa_GetHostError() +{ + return sPaHostError; +} +/************************************************************************************ +** Just count devices so we know how much memory to allocate. +*/ +static BOOL CALLBACK Pa_CountDevProc(LPGUID lpGUID, + LPCTSTR lpszDesc, + LPCTSTR lpszDrvName, + LPVOID lpContext ) +{ + sNumDevices++; + return TRUE; +} +/************************************************************************************ +** Extract capabilities info from each device. +*/ +static BOOL CALLBACK Pa_EnumProc(LPGUID lpGUID, + LPCTSTR lpszDesc, + LPCTSTR lpszDrvName, + LPVOID lpContext ) +{ + HRESULT hr; + LPDIRECTSOUND lpDirectSound; +#if SUPPORT_AUDIO_CAPTURE + LPDIRECTSOUNDCAPTURE lpDirectSoundCapture; +#endif /* SUPPORT_AUDIO_CAPTURE */ + int isInput = (int) lpContext; /* Passed from Pa_CountDevices() */ + internalPortAudioDevice *pad; + + if( sDeviceIndex >= sNumDevices ) + { + sEnumerationError = paInternalError; + return FALSE; + } + pad = &sDevices[sDeviceIndex]; + /* Copy GUID to static array. Set pointer. */ + if( lpGUID == NULL ) + { + pad->pad_lpGUID = NULL; + } + else + { + memcpy( &pad->pad_GUID, lpGUID, sizeof(GUID) ); + pad->pad_lpGUID = &pad->pad_GUID; + } + pad->pad_Info.sampleRates = pad->pad_SampleRates; /* Point to array. */ + /* Allocate room for descriptive name. */ + if( lpszDesc != NULL ) + { + int len = strlen(lpszDesc); + pad->pad_Info.name = (char *)malloc( len+1 ); + if( pad->pad_Info.name == NULL ) + { + sEnumerationError = paInsufficientMemory; + return FALSE; + } + memcpy( (void *) pad->pad_Info.name, lpszDesc, len+1 ); + } +#if SUPPORT_AUDIO_CAPTURE + if( isInput ) + { + /********** Input ******************************/ + DSCCAPS caps; + if( lpGUID == NULL ) sDefaultInputDeviceID = sDeviceIndex; + hr = DirectSoundCaptureCreate( lpGUID, &lpDirectSoundCapture, NULL ); + if( hr != DS_OK ) + { + pad->pad_Info.maxInputChannels = 0; + DBUG(("Cannot create Capture for %s. Result = 0x%x\n", lpszDesc, hr )); + } + else + { + /* Query device characteristics. */ + caps.dwSize = sizeof(caps); + IDirectSoundCapture_GetCaps( lpDirectSoundCapture, &caps ); + /* printf("caps.dwFormats = 0x%x\n", caps.dwFormats ); */ + pad->pad_Info.maxInputChannels = caps.dwChannels; + /* Determine sample rates from flags. */ + if( caps.dwChannels == 2 ) + { + int index = 0; + if( caps.dwFormats & WAVE_FORMAT_1S16) pad->pad_SampleRates[index++] = 11025.0; + if( caps.dwFormats & WAVE_FORMAT_2S16) pad->pad_SampleRates[index++] = 22050.0; + if( caps.dwFormats & WAVE_FORMAT_4S16) pad->pad_SampleRates[index++] = 44100.0; + pad->pad_Info.numSampleRates = index; + } + else if( caps.dwChannels == 1 ) + { + int index = 0; + if( caps.dwFormats & WAVE_FORMAT_1M16) pad->pad_SampleRates[index++] = 11025.0; + if( caps.dwFormats & WAVE_FORMAT_2M16) pad->pad_SampleRates[index++] = 22050.0; + if( caps.dwFormats & WAVE_FORMAT_4M16) pad->pad_SampleRates[index++] = 44100.0; + pad->pad_Info.numSampleRates = index; + } + else pad->pad_Info.numSampleRates = 0; + IDirectSoundCapture_Release( lpDirectSoundCapture ); + } + } + else +#endif /* SUPPORT_AUDIO_CAPTURE */ + + { + /********** Output ******************************/ + DSCAPS caps; + if( lpGUID == NULL ) sDefaultOutputDeviceID = sDeviceIndex; + /* Create interfaces for each object. */ + hr = DirectSoundCreate( lpGUID, &lpDirectSound, NULL ); + if( hr != DS_OK ) + { + pad->pad_Info.maxOutputChannels = 0; + DBUG(("Cannot create dsound for %s. Result = 0x%x\n", lpszDesc, hr )); + } + else + { + /* Query device characteristics. */ + caps.dwSize = sizeof(caps); + IDirectSound_GetCaps( lpDirectSound, &caps ); + pad->pad_Info.maxOutputChannels = ( caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + /* Get sample rates. */ + pad->pad_SampleRates[0] = (double) caps.dwMinSecondarySampleRate; + pad->pad_SampleRates[1] = (double) caps.dwMaxSecondarySampleRate; + if( caps.dwFlags & DSCAPS_CONTINUOUSRATE ) pad->pad_Info.numSampleRates = -1; + else if( caps.dwMinSecondarySampleRate == caps.dwMaxSecondarySampleRate ) + { + if( caps.dwMinSecondarySampleRate == 0 ) + { + /* + ** On my Thinkpad 380Z, DirectSoundV6 returns min-max=0 !! + ** But it supports continuous sampling. + ** So fake range of rates, and hope it really supports it. + */ + pad->pad_SampleRates[0] = 11025.0f; + pad->pad_SampleRates[1] = 48000.0f; + pad->pad_Info.numSampleRates = -1; /* continuous range */ + + DBUG(("PA - Reported rates both zero. Setting to fake values for device #%d\n", sDeviceIndex )); + } + else + { + pad->pad_Info.numSampleRates = 1; + } + } + else if( (caps.dwMinSecondarySampleRate < 1000.0) && (caps.dwMaxSecondarySampleRate > 50000.0) ) + { + /* The EWS88MT drivers lie, lie, lie. The say they only support two rates, 100 & 100000. + ** But we know that they really support a range of rates! + ** So when we see a ridiculous set of rates, assume it is a range. + */ + pad->pad_Info.numSampleRates = -1; + DBUG(("PA - Sample rate range used instead of two odd values for device #%d\n", sDeviceIndex )); + } + else pad->pad_Info.numSampleRates = 2; + IDirectSound_Release( lpDirectSound ); + } + } + pad->pad_Info.nativeSampleFormats = paInt16; + sDeviceIndex++; + return( TRUE ); +} +/*************************************************************************/ +int Pa_CountDevices() +{ + if( sNumDevices <= 0 ) Pa_Initialize(); + return sNumDevices; +} +static internalPortAudioDevice *Pa_GetInternalDevice( PaDeviceID id ) +{ + if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; + return &sDevices[id]; +} +/*************************************************************************/ +const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceID id ) +{ + internalPortAudioDevice *pad; + if( (id < 0) || ( id >= Pa_CountDevices()) ) return NULL; + pad = Pa_GetInternalDevice( id ); + return &pad->pad_Info ; +} +static PaError Pa_MaybeQueryDevices( void ) +{ + if( sNumDevices == 0 ) + { + return Pa_QueryDevices(); + } + return 0; +} +/************************************************************************* +** Returns recommended device ID. +** On the PC, the recommended device can be specified by the user by +** setting an environment variable. For example, to use device #1. +** +** set PA_RECOMMENDED_OUTPUT_DEVICE=1 +** +** The user should first determine the available device ID by using +** the supplied application "pa_devs". +*/ +#define PA_ENV_BUF_SIZE (32) +#define PA_REC_IN_DEV_ENV_NAME ("PA_RECOMMENDED_INPUT_DEVICE") +#define PA_REC_OUT_DEV_ENV_NAME ("PA_RECOMMENDED_OUTPUT_DEVICE") +static PaDeviceID PaHost_GetEnvDefaultDeviceID( char *envName ) +{ + DWORD hresult; + char envbuf[PA_ENV_BUF_SIZE]; + PaDeviceID recommendedID = paNoDevice; + /* Let user determine default device by setting environment variable. */ + hresult = GetEnvironmentVariable( envName, envbuf, PA_ENV_BUF_SIZE ); + if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) + { + recommendedID = atoi( envbuf ); + } + return recommendedID; +} +PaDeviceID Pa_GetDefaultInputDeviceID( void ) +{ + PaError result; + result = PaHost_GetEnvDefaultDeviceID( PA_REC_IN_DEV_ENV_NAME ); + if( result < 0 ) + { + result = Pa_MaybeQueryDevices(); + if( result < 0 ) return result; + result = sDefaultInputDeviceID; + } + return result; +} +PaDeviceID Pa_GetDefaultOutputDeviceID( void ) +{ + PaError result; + result = PaHost_GetEnvDefaultDeviceID( PA_REC_OUT_DEV_ENV_NAME ); + if( result < 0 ) + { + result = Pa_MaybeQueryDevices(); + if( result < 0 ) return result; + result = sDefaultOutputDeviceID; + } + return result; +} +/********************************************************************** +** Make sure that we have queried the device capabilities. +*/ +PaError PaHost_Init( void ) +{ +#if PA_SIMULATE_UNDERFLOW + PRINT(("WARNING - Underflow Simulation Enabled - Expect a Big Glitch!!!\n")); +#endif + return Pa_MaybeQueryDevices(); +} +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; +} +/*******************************************************************/ +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; + } + } +} +/*******************************************************************/ +PaError PaHost_OpenStream( internalPortAudioStream *past ) +{ + HRESULT hr; + PaError result = paNoError; + PaHostSoundControl *pahsc; + int numBytes, maxChannels; + unsigned int minNumBuffers; + internalPortAudioDevice *pad; + DSoundWrapper *dsw; + /* Allocate and initialize host data. */ + pahsc = (PaHostSoundControl *) PaHost_AllocateFastMemory(sizeof(PaHostSoundControl)); /* MEM */ + if( pahsc == NULL ) + { + result = paInsufficientMemory; + goto error; + } + memset( pahsc, 0, sizeof(PaHostSoundControl) ); + past->past_DeviceData = (void *) pahsc; + pahsc->pahsc_TimerID = 0; + dsw = &pahsc->pahsc_DSoundWrapper; + DSW_Init( dsw ); + /* Allocate native buffer. */ + maxChannels = ( past->past_NumOutputChannels > past->past_NumInputChannels ) ? + past->past_NumOutputChannels : past->past_NumInputChannels; + pahsc->pahsc_BytesPerBuffer = past->past_FramesPerUserBuffer * maxChannels * sizeof(short); + if( maxChannels > 0 ) + { + pahsc->pahsc_NativeBuffer = (short *) PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerBuffer); /* MEM */ + if( pahsc->pahsc_NativeBuffer == NULL ) + { + result = paInsufficientMemory; + goto error; + } + } + else + { + result = paInvalidChannelCount; + goto error; + } + + DBUG(("PaHost_OpenStream: pahsc_MinFramesPerHostBuffer = %d\n", pahsc->pahsc_MinFramesPerHostBuffer )); + minNumBuffers = Pa_GetMinNumBuffers( past->past_FramesPerUserBuffer, past->past_SampleRate ); + past->past_NumUserBuffers = ( minNumBuffers > past->past_NumUserBuffers ) ? minNumBuffers : past->past_NumUserBuffers; + numBytes = pahsc->pahsc_BytesPerBuffer * past->past_NumUserBuffers; + if( numBytes < DSBSIZE_MIN ) + { + result = paBufferTooSmall; + goto error; + } + if( numBytes > DSBSIZE_MAX ) + { + result = paBufferTooBig; + goto error; + } + pahsc->pahsc_FramesPerDSBuffer = past->past_FramesPerUserBuffer * past->past_NumUserBuffers; + { + int msecLatency = (int) ((pahsc->pahsc_FramesPerDSBuffer * 1000) / past->past_SampleRate); + PRINT(("PortAudio on DirectSound - Latency = %d frames, %d msec\n", pahsc->pahsc_FramesPerDSBuffer, msecLatency )); + } + /* ------------------ OUTPUT */ + if( (past->past_OutputDeviceID >= 0) && (past->past_NumOutputChannels > 0) ) + { + DBUG(("PaHost_OpenStream: deviceID = 0x%x\n", past->past_OutputDeviceID)); + pad = Pa_GetInternalDevice( past->past_OutputDeviceID ); + hr = DirectSoundCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSound, NULL ); + /* If this fails, then try each output device until we find one that works. */ + if( hr != DS_OK ) + { + int i; + ERR_RPT(("Creation of requested Audio Output device '%s' failed.\n", + ((pad->pad_lpGUID == NULL) ? "Default" : pad->pad_Info.name) )); + for( i=0; i<Pa_CountDevices(); i++ ) + { + pad = Pa_GetInternalDevice( i ); + if( pad->pad_Info.maxOutputChannels >= past->past_NumOutputChannels ) + { + DBUG(("Try device '%s' instead.\n", pad->pad_Info.name )); + hr = DirectSoundCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSound, NULL ); + if( hr == DS_OK ) + { + ERR_RPT(("Using device '%s' instead.\n", pad->pad_Info.name )); + break; + } + } + } + } + if( hr != DS_OK ) + { + ERR_RPT(("PortAudio: DirectSoundCreate() failed!\n")); + result = paHostError; + sPaHostError = hr; + goto error; + } + hr = DSW_InitOutputBuffer( dsw, + (unsigned long) (past->past_SampleRate + 0.5), + past->past_NumOutputChannels, numBytes ); + DBUG(("DSW_InitOutputBuffer() returns %x\n", hr)); + if( hr != DS_OK ) + { + result = paHostError; + sPaHostError = hr; + goto error; + } + past->past_FrameCount = pahsc->pahsc_DSoundWrapper.dsw_FramesWritten; + } +#if SUPPORT_AUDIO_CAPTURE + /* ------------------ INPUT */ + if( (past->past_InputDeviceID >= 0) && (past->past_NumInputChannels > 0) ) + { + pad = Pa_GetInternalDevice( past->past_InputDeviceID ); + hr = DirectSoundCaptureCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); + /* If this fails, then try each input device until we find one that works. */ + if( hr != DS_OK ) + { + int i; + ERR_RPT(("Creation of requested Audio Capture device '%s' failed.\n", + ((pad->pad_lpGUID == NULL) ? "Default" : pad->pad_Info.name) )); + for( i=0; i<Pa_CountDevices(); i++ ) + { + pad = Pa_GetInternalDevice( i ); + if( pad->pad_Info.maxInputChannels >= past->past_NumInputChannels ) + { + PRINT(("Try device '%s' instead.\n", pad->pad_Info.name )); + hr = DirectSoundCaptureCreate( pad->pad_lpGUID, &dsw->dsw_pDirectSoundCapture, NULL ); + if( hr == DS_OK ) break; + } + } + } + if( hr != DS_OK ) + { + ERR_RPT(("PortAudio: DirectSoundCaptureCreate() failed!\n")); + result = paHostError; + sPaHostError = hr; + goto error; + } + hr = DSW_InitInputBuffer( dsw, + (unsigned long) (past->past_SampleRate + 0.5), + past->past_NumInputChannels, numBytes ); + DBUG(("DSW_InitInputBuffer() returns %x\n", hr)); + if( hr != DS_OK ) + { + ERR_RPT(("PortAudio: DSW_InitInputBuffer() returns %x\n", hr)); + result = paHostError; + sPaHostError = hr; + goto error; + } + } +#endif /* SUPPORT_AUDIO_CAPTURE */ + /* Calculate scalar used in CPULoad calculation. */ + { + LARGE_INTEGER frequency; + if( QueryPerformanceFrequency( &frequency ) == 0 ) + { + pahsc->pahsc_InverseTicksPerUserBuffer = 0.0; + } + else + { + pahsc->pahsc_InverseTicksPerUserBuffer = past->past_SampleRate / + ( (double)frequency.QuadPart * past->past_FramesPerUserBuffer ); + DBUG(("pahsc_InverseTicksPerUserBuffer = %g\n", pahsc->pahsc_InverseTicksPerUserBuffer )); + } + } + return result; +error: + PaHost_CloseStream( past ); + return result; +} +/*************************************************************************/ +PaError PaHost_StartOutput( internalPortAudioStream *past ) +{ + HRESULT hr; + PaHostSoundControl *pahsc; + PaError result = paNoError; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + /* Give user callback a chance to pre-fill buffer. */ + result = Pa_TimeSlice( past ); + if( result != paNoError ) return result; // FIXME - what if finished? + hr = DSW_StartOutput( &pahsc->pahsc_DSoundWrapper ); + DBUG(("PaHost_StartOutput: DSW_StartOutput returned = 0x%X.\n", hr)); + if( hr != DS_OK ) + { + result = paHostError; + sPaHostError = hr; + goto error; + } +error: + return result; +} +/*************************************************************************/ +PaError PaHost_StartInput( internalPortAudioStream *past ) +{ + PaError result = paNoError; +#if SUPPORT_AUDIO_CAPTURE + HRESULT hr; + PaHostSoundControl *pahsc; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + hr = DSW_StartInput( &pahsc->pahsc_DSoundWrapper ); + DBUG(("Pa_StartStream: DSW_StartInput returned = 0x%X.\n", hr)); + if( hr != DS_OK ) + { + result = paHostError; + sPaHostError = hr; + goto error; + } +error: +#endif /* SUPPORT_AUDIO_CAPTURE */ + return result; +} +/*************************************************************************/ +PaError PaHost_StartEngine( internalPortAudioStream *past ) +{ + PaHostSoundControl *pahsc; + PaError result = paNoError; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + past->past_StopNow = 0; + past->past_StopSoon = 0; + past->past_IsActive = 1; + /* Create timer that will wake us up so we can fill the DSound buffer. */ + { + int msecPerBuffer; + int resolution; + int bufsPerInterrupt; + + DBUG(("PaHost_StartEngine: past_NumUserBuffers = %d\n", past->past_NumUserBuffers)); + /* Decide how often to wake up and fill the buffers. */ + if( past->past_NumUserBuffers == 2 ) + { + /* Generate two timer interrupts per user buffer. */ + msecPerBuffer = (500 * past->past_FramesPerUserBuffer) / (int) past->past_SampleRate; + } + else + { + if ( past->past_NumUserBuffers >= 16 ) bufsPerInterrupt = past->past_NumUserBuffers/8; + else if ( past->past_NumUserBuffers >= 8 ) bufsPerInterrupt = 2; + else bufsPerInterrupt = 1; + + msecPerBuffer = 1000 * (bufsPerInterrupt * past->past_FramesPerUserBuffer) / (int) past->past_SampleRate; + + DBUG(("PaHost_StartEngine: bufsPerInterrupt = %d\n", bufsPerInterrupt)); + } + + DBUG(("PaHost_StartEngine: msecPerBuffer = %d\n", msecPerBuffer)); + + if( msecPerBuffer < 10 ) msecPerBuffer = 10; + else if( msecPerBuffer > 100 ) msecPerBuffer = 100; + DBUG(("PaHost_StartEngine: clipped msecPerBuffer = %d\n", msecPerBuffer)); + + resolution = msecPerBuffer/4; + pahsc->pahsc_TimerID = timeSetEvent( msecPerBuffer, resolution, (LPTIMECALLBACK) Pa_TimerCallback, + (DWORD) past, TIME_PERIODIC ); + } + if( pahsc->pahsc_TimerID == 0 ) + { + past->past_IsActive = 0; + result = paHostError; + sPaHostError = 0; + goto error; + } +error: + return result; +} +/*************************************************************************/ +PaError PaHost_StopEngine( internalPortAudioStream *past, int abort ) +{ + int timeoutMsec; + PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return paNoError; + if( abort ) past->past_StopNow = 1; + past->past_StopSoon = 1; + /* Set timeout at 20% beyond maximum time we might wait. */ + timeoutMsec = (int) (1200.0 * pahsc->pahsc_FramesPerDSBuffer / past->past_SampleRate); + while( past->past_IsActive && (timeoutMsec > 0) ) + { + Sleep(10); + timeoutMsec -= 10; + } + if( pahsc->pahsc_TimerID != 0 ) + { + timeKillEvent(pahsc->pahsc_TimerID); /* Stop callback timer. */ + pahsc->pahsc_TimerID = 0; + } + return paNoError; +} +/*************************************************************************/ +PaError PaHost_StopInput( internalPortAudioStream *past, int abort ) +{ +#if SUPPORT_AUDIO_CAPTURE + HRESULT hr; + PaHostSoundControl *pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return paNoError; + (void) abort; + hr = DSW_StopInput( &pahsc->pahsc_DSoundWrapper ); + DBUG(("DSW_StopInput() result is %x\n", hr)); +#endif /* SUPPORT_AUDIO_CAPTURE */ + return paNoError; +} +/*************************************************************************/ +PaError PaHost_StopOutput( internalPortAudioStream *past, int abort ) +{ + HRESULT hr; + PaHostSoundControl *pahsc; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return paNoError; + (void) abort; + hr = DSW_StopOutput( &pahsc->pahsc_DSoundWrapper ); + DBUG(("DSW_StopOutput() result is %x\n", hr)); + return paNoError; +} +/*******************************************************************/ +PaError PaHost_CloseStream( internalPortAudioStream *past ) +{ + PaHostSoundControl *pahsc; + if( past == NULL ) return paBadStreamPtr; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return paNoError; + DSW_Term( &pahsc->pahsc_DSoundWrapper ); + if( pahsc->pahsc_NativeBuffer ) + { + PaHost_FreeFastMemory( pahsc->pahsc_NativeBuffer, pahsc->pahsc_BytesPerBuffer ); /* MEM */ + pahsc->pahsc_NativeBuffer = NULL; + } + PaHost_FreeFastMemory( pahsc, sizeof(PaHostSoundControl) ); /* MEM */ + past->past_DeviceData = NULL; + return paNoError; +} + +/* Set minimal latency based on whether NT or Win95. + * NT has higher latency. + */ +static int PaHost_GetMinSystemLatency( void ) +{ + int minLatencyMsec; + /* Set minimal latency based on whether NT or other OS. + * NT has higher latency. + */ + OSVERSIONINFO osvi; + osvi.dwOSVersionInfoSize = sizeof( osvi ); + GetVersionEx( &osvi ); + DBUG(("PA - PlatformId = 0x%x\n", osvi.dwPlatformId )); + DBUG(("PA - MajorVersion = 0x%x\n", osvi.dwMajorVersion )); + DBUG(("PA - MinorVersion = 0x%x\n", osvi.dwMinorVersion )); + /* Check for NT */ + if( (osvi.dwMajorVersion == 4) && (osvi.dwPlatformId == 2) ) + { + minLatencyMsec = PA_WIN_NT_LATENCY; + } + else if(osvi.dwMajorVersion >= 5) + { + minLatencyMsec = PA_WIN_WDM_LATENCY; + } + else + { + minLatencyMsec = PA_WIN_9X_LATENCY; + } + return minLatencyMsec; +} + +/************************************************************************* +** Determine minimum number of buffers required for this host based +** on minimum latency. Latency can be optionally set by user by setting +** an environment variable. For example, to set latency to 200 msec, put: +** +** set PA_MIN_LATENCY_MSEC=200 +** +** in the AUTOEXEC.BAT file and reboot. +** If the environment variable is not set, then the latency will be determined +** based on the OS. Windows NT has higher latency than Win95. +*/ +#define PA_LATENCY_ENV_NAME ("PA_MIN_LATENCY_MSEC") +int Pa_GetMinNumBuffers( int framesPerBuffer, double sampleRate ) +{ + char envbuf[PA_ENV_BUF_SIZE]; + DWORD hresult; + int minLatencyMsec = 0; + double msecPerBuffer = (1000.0 * framesPerBuffer) / sampleRate; + int minBuffers; + /* Let user determine minimal latency by setting environment variable. */ + hresult = GetEnvironmentVariable( PA_LATENCY_ENV_NAME, envbuf, PA_ENV_BUF_SIZE ); + if( (hresult > 0) && (hresult < PA_ENV_BUF_SIZE) ) + { + minLatencyMsec = atoi( envbuf ); + } + else + { + minLatencyMsec = PaHost_GetMinSystemLatency(); +#if PA_USE_HIGH_LATENCY + PRINT(("PA - Minimum Latency set to %d msec!\n", minLatencyMsec )); +#endif + + } + minBuffers = (int) (1.0 + ((double)minLatencyMsec / msecPerBuffer)); + if( minBuffers < 2 ) minBuffers = 2; + return minBuffers; +} +/*************************************************************************/ +PaError PaHost_Term( void ) +{ + int i; + /* Free names allocated during enumeration. */ + for( i=0; i<sNumDevices; i++ ) + { + if( sDevices[i].pad_Info.name != NULL ) + { + free( (void *) sDevices[i].pad_Info.name ); + sDevices[i].pad_Info.name = NULL; + } + } + if( sDevices != NULL ) + { + PaHost_FreeFastMemory( sDevices, sNumDevices * sizeof(internalPortAudioDevice) ); /* MEM */ + sDevices = NULL; + sNumDevices = 0; + } + return 0; +} +void Pa_Sleep( long msec ) +{ + Sleep( msec ); +} +/************************************************************************* + * Allocate memory that can be accessed in real-time. + * This may need to be held in physical memory so that it is not + * paged to virtual memory. + * This call MUST be balanced with a call to PaHost_FreeFastMemory(). + * Memory will be set to zero. + */ +void *PaHost_AllocateFastMemory( long numBytes ) +{ + void *addr = GlobalAlloc( GPTR, numBytes ); /* FIXME - do we need physical memory? Use VirtualLock() */ /* MEM */ + return addr; +} +/************************************************************************* + * Free memory that could be accessed in real-time. + * This call MUST be balanced with a call to PaHost_AllocateFastMemory(). + */ +void PaHost_FreeFastMemory( void *addr, long numBytes ) +{ + if( addr != NULL ) GlobalFree( addr ); /* MEM */ +} +/***********************************************************************/ +PaError PaHost_StreamActive( internalPortAudioStream *past ) +{ + PaHostSoundControl *pahsc; + if( past == NULL ) return paBadStreamPtr; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + if( pahsc == NULL ) return paInternalError; + return (PaError) (past->past_IsActive); +} +/*************************************************************************/ +PaTimestamp Pa_StreamTime( PortAudioStream *stream ) +{ + DSoundWrapper *dsw; + internalPortAudioStream *past = (internalPortAudioStream *) stream; + PaHostSoundControl *pahsc; + if( past == NULL ) return paBadStreamPtr; + pahsc = (PaHostSoundControl *) past->past_DeviceData; + dsw = &pahsc->pahsc_DSoundWrapper; + return dsw->dsw_FramesPlayed; +} diff --git a/pd/portaudio_v18/pa_win_ds/portaudio.def b/pd/portaudio_v18/pa_win_ds/portaudio.def new file mode 100644 index 00000000..8012b99e --- /dev/null +++ b/pd/portaudio_v18/pa_win_ds/portaudio.def @@ -0,0 +1,28 @@ +LIBRARY PortAudio +DESCRIPTION 'PortAudio Portable interface to audio HW' + +EXPORTS + ; Explicit exports can go here + Pa_Initialize @1 + Pa_Terminate @2 + Pa_GetHostError @3 + Pa_GetErrorText @4 + Pa_CountDevices @5 + Pa_GetDefaultInputDeviceID @6 + Pa_GetDefaultOutputDeviceID @7 + Pa_GetDeviceInfo @8 + Pa_OpenStream @9 + Pa_OpenDefaultStream @10 + Pa_CloseStream @11 + Pa_StartStream @12 + Pa_StopStream @13 + Pa_StreamActive @14 + Pa_StreamTime @15 + Pa_GetCPULoad @16 + Pa_GetMinNumBuffers @17 + Pa_Sleep @18 + + ;123456789012345678901234567890123456 + ;000000000111111111122222222223333333 + + |