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-rw-r--r--pd/portaudio_v18/pa_win_ds/dsound_wrapper.c466
-rw-r--r--pd/portaudio_v18/pa_win_ds/dsound_wrapper.h106
-rw-r--r--pd/portaudio_v18/pa_win_ds/pa_dsound.c1042
-rw-r--r--pd/portaudio_v18/pa_win_ds/portaudio.def28
4 files changed, 0 insertions, 1642 deletions
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 <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( &currentTime );
- 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 <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
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 <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
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
-
-
generated by cgit v1.2.1 (git 2.18.0) at 2024-06-26 18:44:52 +0000