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373 lines
12 KiB
C
373 lines
12 KiB
C
/*
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* Soundblaster Emulation
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*
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* Copyright 2002 Christian Costa
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
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*/
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#include "config.h"
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#include <stdarg.h>
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#include "windef.h"
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#include "winbase.h"
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#include "dosexe.h"
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#include "wine/debug.h"
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#include "wingdi.h"
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#include "mmsystem.h"
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#include "dsound.h"
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WINE_DEFAULT_DEBUG_CHANNEL(sblaster);
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/* Board Configuration */
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/* FIXME: Should be in a config file */
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#define SB_IRQ 5
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#define SB_IRQ_PRI 11
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#define SB_DMA 1
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/* Soundblaster state */
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static int SampleMode; /* Mono / Stereo */
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static int SampleRate;
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static int SamplesCount;
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static BYTE DSP_Command[256]; /* Store param numbers in bytes for each command */
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static BYTE DSP_InBuffer[10]; /* Store DSP command bytes parameters from host */
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static int InSize; /* Nb of bytes in InBuffer */
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static BYTE DSP_OutBuffer[10]; /* Store DSP information bytes to host */
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static int OutSize; /* Nb of bytes in InBuffer */
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static int command; /* Current command */
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static BOOL end_sound_loop = FALSE;
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static BOOL dma_enable = FALSE;
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/* The maximum size of a dma transfer can be 65536 */
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#define DMATRFSIZE 1024
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/* DMA can perform 8 or 16-bit transfer */
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static BYTE dma_buffer[DMATRFSIZE*2];
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/* Direct Sound buffer config */
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#define DSBUFLEN 4096 /* FIXME: Only this value seems to work */
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/* Direct Sound playback stuff */
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static LPDIRECTSOUND lpdsound;
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static LPDIRECTSOUNDBUFFER lpdsbuf;
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static DSBUFFERDESC buf_desc;
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static WAVEFORMATEX wav_fmt;
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static HANDLE SB_Thread;
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static UINT buf_off;
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extern HWND vga_hwnd;
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/* SB_Poll performs DMA transfers and fills the Direct Sound Buffer */
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static DWORD CALLBACK SB_Poll( void *dummy )
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{
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HRESULT result;
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LPBYTE lpbuf1 = NULL;
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LPBYTE lpbuf2 = NULL;
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DWORD dwsize1 = 0;
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DWORD dwsize2 = 0;
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DWORD dwbyteswritten1 = 0;
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DWORD dwbyteswritten2 = 0;
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int size;
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/* FIXME: this loop must be improved */
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while(!end_sound_loop)
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{
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Sleep(10);
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if (dma_enable) {
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size = DMA_Transfer(SB_DMA,min(DMATRFSIZE,SamplesCount),dma_buffer);
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} else
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continue;
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result = IDirectSoundBuffer_Lock(lpdsbuf,buf_off,size,(LPVOID *)&lpbuf1,&dwsize1,(LPVOID *)&lpbuf2,&dwsize2,0);
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if (result != DS_OK) {
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ERR("Unable to lock sound buffer !\n");
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continue;
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}
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dwbyteswritten1 = min(size,dwsize1);
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memcpy(lpbuf1,dma_buffer,dwbyteswritten1);
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if (size>dwsize1) {
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dwbyteswritten2 = min(size - dwbyteswritten1,dwsize2);
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memcpy(lpbuf2,dma_buffer+dwbyteswritten1,dwbyteswritten2);
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}
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buf_off = (buf_off + dwbyteswritten1 + dwbyteswritten2) % DSBUFLEN;
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result = IDirectSoundBuffer_Unlock(lpdsbuf,lpbuf1,dwbyteswritten1,lpbuf2,dwbyteswritten2);
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if (result!=DS_OK)
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ERR("Unable to unlock sound buffer !\n");
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SamplesCount -= size;
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if (!SamplesCount) dma_enable = FALSE;
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}
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return 0;
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}
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static BOOL SB_Init(void)
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{
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HRESULT result;
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if (!lpdsound) {
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result = DirectSoundCreate(NULL,&lpdsound,NULL);
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if (result != DS_OK) {
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ERR("Unable to initialize Sound Subsystem err = %x !\n",result);
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return FALSE;
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}
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/* FIXME: To uncomment when :
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- SetCooperative level is correctly implemented
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- an always valid and non changing handle to a windows (vga_hwnd) is available
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(this surely needs some work in vga.c)
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result = IDirectSound_SetCooperativeLevel(lpdsound,vga_hwnd,DSSCL_EXCLUSIVE|DSSCL_PRIORITY);
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if (result != DS_OK) {
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ERR("Can't set cooperative level !\n");
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return FALSE;
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}
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*/
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/* Default format */
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wav_fmt.wFormatTag = WAVE_FORMAT_PCM;
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wav_fmt.nChannels = 1;
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wav_fmt.nSamplesPerSec = 22050;
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wav_fmt.nAvgBytesPerSec = 22050;
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wav_fmt.nBlockAlign = 1;
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wav_fmt.wBitsPerSample = 8;
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wav_fmt.cbSize = 0;
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memset(&buf_desc,0,sizeof(DSBUFFERDESC));
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buf_desc.dwSize = sizeof(DSBUFFERDESC);
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buf_desc.dwBufferBytes = DSBUFLEN;
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buf_desc.lpwfxFormat = &wav_fmt;
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result = IDirectSound_CreateSoundBuffer(lpdsound,&buf_desc,&lpdsbuf,NULL);
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if (result != DS_OK) {
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ERR("Can't create sound buffer !\n");
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return FALSE;
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}
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result = IDirectSoundBuffer_Play(lpdsbuf,0, 0, DSBPLAY_LOOPING);
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if (result != DS_OK) {
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ERR("Can't start playing !\n");
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return FALSE;
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}
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buf_off = 0;
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end_sound_loop = FALSE;
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SB_Thread = CreateThread(NULL, 0, SB_Poll, NULL, 0, NULL);
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TRACE("thread\n");
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if (!SB_Thread) {
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ERR("Can't create thread !\n");
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return FALSE;
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}
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}
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return TRUE;
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}
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static void SB_Reset(void)
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{
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int i;
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for(i=0;i<256;i++)
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DSP_Command[i]=0;
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/* Set Time Constant */
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DSP_Command[0x40]=1;
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/* Generate IRQ */
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DSP_Command[0xF2]=0;
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/* DMA DAC 8-bits */
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DSP_Command[0x14]=2;
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/* Generic DAC/ADC DMA (16-bit, 8-bit) */
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for(i=0xB0;i<=0xCF;i++)
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DSP_Command[i]=3;
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/* DSP Identification */
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DSP_Command[0xE0]=1;
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/* Clear command and input buffer */
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command = -1;
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InSize = 0;
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/* Put a garbage value in the output buffer */
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OutSize = 1;
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if (SB_Init())
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/* All right, let's put the magic value for autodetection */
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DSP_OutBuffer[0] = 0xaa;
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else
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/* Something is wrong, put 0 to failed autodetection */
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DSP_OutBuffer[0] = 0x00;
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}
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/* Find a standard sampling rate for DirectSound */
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static int SB_StdSampleRate(int SampleRate)
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{
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if (SampleRate>((44100+48000)/2)) return 48000;
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if (SampleRate>((32000+44100)/2)) return 44100;
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if (SampleRate>((24000+32000)/2)) return 32000;
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if (SampleRate>((22050+24000)/2)) return 24000;
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if (SampleRate>((16000+22050)/2)) return 22050;
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if (SampleRate>((12000+16000)/2)) return 16000;
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if (SampleRate>((11025+12000)/2)) return 12000;
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if (SampleRate>((8000+11025)/2)) return 11025;
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return 8000;
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}
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void SB_ioport_out( WORD port, BYTE val )
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{
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switch(port)
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{
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/* DSP - Reset */
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case 0x226:
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TRACE("Resetting DSP.\n");
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SB_Reset();
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break;
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/* DSP - Write Data or Command */
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case 0x22c:
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TRACE("val=%x\n",val);
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if (command == -1) {
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/* Clear input buffer and set the current command */
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command = val;
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InSize = 0;
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}
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if (InSize!=DSP_Command[command])
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/* Fill the input buffer the command parameters if any */
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DSP_InBuffer[InSize++]=val;
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else {
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/* Process command */
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switch(command)
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{
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case 0x10: /* SB */
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FIXME("Direct DAC (8-bit) - Not Implemented\n");
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break;
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case 0x14: /* SB */
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SamplesCount = DSP_InBuffer[1]+(val<<8)+1;
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TRACE("DMA DAC (8-bit) for %x samples\n",SamplesCount);
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dma_enable = TRUE;
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break;
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case 0x20:
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FIXME("Direct ADC (8-bit) - Not Implemented\n");
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break;
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case 0x24: /* SB */
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FIXME("DMA ADC (8-bit) - Not Implemented\n");
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break;
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case 0x40: /* SB */
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SampleRate = 1000000/(256-val);
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TRACE("Set Time Constant (%d <-> %d Hz => %d Hz)\n",DSP_InBuffer[0],
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SampleRate,SB_StdSampleRate(SampleRate));
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SampleRate = SB_StdSampleRate(SampleRate);
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wav_fmt.nSamplesPerSec = SampleRate;
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wav_fmt.nAvgBytesPerSec = SampleRate;
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IDirectSoundBuffer_SetFormat(lpdsbuf,&wav_fmt);
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break;
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/* case 0xBX/0xCX -> See below */
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case 0xD0: /* SB */
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TRACE("Halt DMA operation (8-bit)\n");
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dma_enable = FALSE;
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break;
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case 0xD1: /* SB */
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FIXME("Enable Speaker - Not Implemented\n");
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break;
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case 0xD3: /* SB */
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FIXME("Disable Speaker - Not Implemented\n");
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break;
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case 0xD4: /* SB */
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FIXME("Continue DMA operation (8-bit) - Not Implemented\n");
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break;
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case 0xD8: /* SB */
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FIXME("Speaker Status - Not Implemented\n");
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break;
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case 0xE0: /* SB 2.0 */
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TRACE("DSP Identification\n");
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DSP_OutBuffer[OutSize++] = ~val;
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break;
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case 0xE1: /* SB */
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TRACE("DSP Version\n");
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OutSize=2;
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DSP_OutBuffer[0]=0; /* returns version 1.0 */
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DSP_OutBuffer[1]=1;
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break;
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case 0xF2: /* SB */
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TRACE("IRQ Request (8-bit)\n");
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break;
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default:
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if (((command&0xF0)==0xB0)||((DSP_InBuffer[0]&0xF0)==0xC0)) {
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/* SB16 */
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FIXME("Generic DAC/ADC DMA (16-bit, 8-bit) - %d % d\n",command,DSP_InBuffer[1]);
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if (command&0x02)
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FIXME("Generic DAC/ADC fifo mode not supported\n");
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if (command&0x04)
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FIXME("Generic DAC/ADC autoinit dma mode not supported\n");
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if (command&0x08)
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FIXME("Generic DAC/ADC adc mode not supported\n");
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switch(command>>4) {
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case 0xB:
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FIXME("Generic DAC/ADC 8-bit not supported\n");
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SampleMode = 0;
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break;
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case 0xC:
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FIXME("Generic DAC/ADC 16-bit not supported\n");
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SampleMode = 1;
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break;
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default:
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ERR("Generic DAC/ADC resolution unknown\n");
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break;
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}
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if (DSP_InBuffer[1]&0x010)
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FIXME("Generic DAC/ADC signed sample mode not supported\n");
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if (DSP_InBuffer[1]&0x020)
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FIXME("Generic DAC/ADC stereo mode not supported\n");
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SamplesCount = DSP_InBuffer[2]+(val<<8)+1;
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TRACE("Generic DMA for %x samples\n",SamplesCount);
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dma_enable = TRUE;
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}
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else
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FIXME("DSP command %x not supported\n",val);
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}
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/* Empty the input buffer and end the command */
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InSize = 0;
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command = -1;
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}
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}
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}
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BYTE SB_ioport_in( WORD port )
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{
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BYTE res = 0;
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switch(port)
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{
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/* DSP Read Data */
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case 0x22a:
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/* Value in the read buffer */
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if (OutSize)
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res = DSP_OutBuffer[--OutSize];
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else
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/* return the last byte */
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res = DSP_OutBuffer[0];
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break;
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/* DSP - Write Buffer Status */
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case 0x22c:
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/* DSP always ready for writing */
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res = 0x00;
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break;
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/* DSP - Data Available Status */
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/* DSP - IRQ Acknowledge, 8-bit */
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case 0x22e:
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/* DSP data availability check */
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if (OutSize)
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res = 0x80;
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else
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res = 0x00;
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break;
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}
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return res;
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}
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