From fe8aa4ce5e8eebc1c6f762f4fc40328718a13e22 Mon Sep 17 00:00:00 2001 From: Miller Puckette Date: Sat, 31 Dec 2005 01:32:12 +0000 Subject: Deleted unused (?) files svn path=/trunk/; revision=4318 --- pd/portaudio_v18/pa_win_ds/dsound_wrapper.c | 466 ------------ pd/portaudio_v18/pa_win_ds/dsound_wrapper.h | 106 --- pd/portaudio_v18/pa_win_ds/pa_dsound.c | 1042 --------------------------- pd/portaudio_v18/pa_win_ds/portaudio.def | 28 - 4 files changed, 1642 deletions(-) delete mode 100644 pd/portaudio_v18/pa_win_ds/dsound_wrapper.c delete mode 100644 pd/portaudio_v18/pa_win_ds/dsound_wrapper.h delete mode 100644 pd/portaudio_v18/pa_win_ds/pa_dsound.c delete mode 100644 pd/portaudio_v18/pa_win_ds/portaudio.def (limited to 'pd/portaudio_v18/pa_win_ds') diff --git a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c b/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c deleted file mode 100644 index 527f0eb3..00000000 --- a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.c +++ /dev/null @@ -1,466 +0,0 @@ -/* - * $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 -#include -#include -#define INITGUID // Needed to build IID_IDirectSoundNotify. See objbase.h for info. -#include -#include -#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 deleted file mode 100644 index b04b0020..00000000 --- a/pd/portaudio_v18/pa_win_ds/dsound_wrapper.h +++ /dev/null @@ -1,106 +0,0 @@ -#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 -#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 deleted file mode 100644 index 52532625..00000000 --- a/pd/portaudio_v18/pa_win_ds/pa_dsound.c +++ /dev/null @@ -1,1042 +0,0 @@ -/* - * $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 -#include -#ifndef __MWERKS__ -#include -#include -#endif //__MWERKS__ -#include -#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; ipad_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; ipad_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; ipast_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 deleted file mode 100644 index 8012b99e..00000000 --- a/pd/portaudio_v18/pa_win_ds/portaudio.def +++ /dev/null @@ -1,28 +0,0 @@ -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 - - -- cgit v1.2.1