Sweden-Number/dlls/winedos/soundblaster.c

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/*
* Soundblaster Emulation
*
* Copyright 2002 Christian Costa
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "config.h"
#include <stdarg.h>
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#include "windef.h"
#include "winbase.h"
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#include "dosexe.h"
#include "wine/debug.h"
#include "wingdi.h"
#include "mmsystem.h"
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#include "dsound.h"
WINE_DEFAULT_DEBUG_CHANNEL(sblaster);
/* Board Configuration */
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/* FIXME: Should be in a config file */
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#define SB_IRQ 5
#define SB_IRQ_PRI 11
#define SB_DMA 1
/* Soundblaster state */
static int SampleMode; /* Mono / Stereo */
static int SampleRate;
static int SamplesCount;
static BYTE DSP_Command[256]; /* Store param numbers in bytes for each command */
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 */
static int end_sound_loop = 0;
static int dma_enable = 0;
/* 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 */
static BYTE dma_buffer[DMATRFSIZE*2];
/* Direct Sound buffer config */
#define DSBUFLEN 4096 /* FIXME: Only this value seems to work */
/* Direct Sound playback stuff */
static HMODULE hmodule;
typedef HRESULT (WINAPI* fnDirectSoundCreate) (LPGUID,LPDIRECTSOUND*,LPUNKNOWN);
fnDirectSoundCreate lpDirectSoundCreate;
static LPDIRECTSOUND lpdsound;
static LPDIRECTSOUNDBUFFER lpdsbuf;
static DSBUFFERDESC buf_desc;
static WAVEFORMATEX wav_fmt;
static HANDLE SB_Thread;
static UINT buf_off;
extern HWND vga_hwnd;
/* SB_Poll performs DMA transfers and fills the Direct Sound Buffer */
static DWORD CALLBACK SB_Poll( void *dummy )
{
HRESULT result;
LPBYTE lpbuf1 = NULL;
LPBYTE lpbuf2 = NULL;
DWORD dwsize1 = 0;
DWORD dwsize2 = 0;
DWORD dwbyteswritten1 = 0;
DWORD dwbyteswritten2 = 0;
int size;
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/* FIXME: this loop must be improved */
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while(!end_sound_loop)
{
Sleep(10);
if (dma_enable) {
size = DMA_Transfer(SB_DMA,min(DMATRFSIZE,SamplesCount),dma_buffer);
} else
continue;
result = IDirectSoundBuffer_Lock(lpdsbuf,buf_off,size,&lpbuf1,&dwsize1,&lpbuf2,&dwsize2,0);
if (result != DS_OK) {
ERR("Unable to lock sound buffer !\n");
continue;
}
dwbyteswritten1 = min(size,dwsize1);
memcpy(lpbuf1,dma_buffer,dwbyteswritten1);
if (size>dwsize1) {
dwbyteswritten2 = min(size - dwbyteswritten1,dwsize2);
memcpy(lpbuf2,dma_buffer+dwbyteswritten1,dwbyteswritten2);
}
buf_off = (buf_off + dwbyteswritten1 + dwbyteswritten2) % DSBUFLEN;
result = IDirectSoundBuffer_Unlock(lpdsbuf,lpbuf1,dwbyteswritten1,lpbuf2,dwbyteswritten2);
if (result!=DS_OK)
ERR("Unable to unlock sound buffer !\n");
SamplesCount -= size;
if (!SamplesCount) {
DOSVM_QueueEvent(SB_IRQ,SB_IRQ_PRI,NULL,NULL);
dma_enable = 0;
}
}
return 0;
}
static BOOL SB_Init(void)
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{
HRESULT result;
if (!lpdsound) {
hmodule = LoadLibraryA("dsound.dll");
if (!hmodule) {
ERR("Can't load dsound.dll !\n");
return 0;
}
lpDirectSoundCreate = (fnDirectSoundCreate)GetProcAddress(hmodule,"DirectSoundCreate");
if (!lpDirectSoundCreate) {
/* CloseHandle(hmodule); */
ERR("Can't find DirectSoundCreate function !\n");
return 0;
}
result = (*lpDirectSoundCreate)(NULL,&lpdsound,NULL);
if (result != DS_OK) {
ERR("Unable to initialize Sound Subsystem err = %lx !\n",result);
return 0;
}
/* FIXME: To uncomment when :
- SetCooperative level is correctly implemented
- an always valid and non changing handle to a windows (vga_hwnd) is available
(this surely needs some work in vga.c)
result = IDirectSound_SetCooperativeLevel(lpdsound,vga_hwnd,DSSCL_EXCLUSIVE|DSSCL_PRIORITY);
if (result != DS_OK) {
ERR("Can't set cooperative level !\n");
return 0;
}
*/
/* Default format */
wav_fmt.wFormatTag = WAVE_FORMAT_PCM;
wav_fmt.nChannels = 1;
wav_fmt.nSamplesPerSec = 22050;
wav_fmt.nAvgBytesPerSec = 22050;
wav_fmt.nBlockAlign = 1;
wav_fmt.wBitsPerSample = 8;
wav_fmt.cbSize = 0;
memset(&buf_desc,0,sizeof(DSBUFFERDESC));
buf_desc.dwSize = sizeof(DSBUFFERDESC);
buf_desc.dwBufferBytes = DSBUFLEN;
buf_desc.lpwfxFormat = &wav_fmt;
result = IDirectSound_CreateSoundBuffer(lpdsound,&buf_desc,&lpdsbuf,NULL);
if (result != DS_OK) {
ERR("Can't create sound buffer !\n");
return 0;
}
result = IDirectSoundBuffer_Play(lpdsbuf,0, 0, DSBPLAY_LOOPING);
if (result != DS_OK) {
ERR("Can't start playing !\n");
return 0;
}
buf_off = 0;
end_sound_loop = 0;
SB_Thread = CreateThread(NULL, 0, SB_Poll, NULL, 0, NULL);
TRACE("thread\n");
if (!SB_Thread) {
ERR("Can't create thread !\n");
return 0;
}
}
return 1;
}
static void SB_Reset(void)
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{
int i;
for(i=0;i<256;i++)
DSP_Command[i]=0;
/* Set Time Constant */
DSP_Command[0x40]=1;
/* Generate IRQ */
DSP_Command[0xF2]=0;
/* DMA DAC 8-bits */
DSP_Command[0x14]=2;
/* Generic DAC/ADC DMA (16-bit, 8-bit) */
for(i=0xB0;i<=0xCF;i++)
DSP_Command[i]=3;
/* DSP Indentification */
DSP_Command[0xE0]=1;
/* Clear command and input buffer */
command = -1;
InSize = 0;
/* Put a garbage value in the output buffer */
OutSize = 1;
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;
else
/* Something is wrong, put 0 to failed audetection */
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DSP_OutBuffer[0] = 0x00;
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}
/* Find a standard sampling rate for DirectSound */
static int SB_StdSampleRate(int SampleRate)
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{
if (SampleRate>((44100+48000)/2)) return 48000;
if (SampleRate>((32000+44100)/2)) return 44100;
if (SampleRate>((24000+32000)/2)) return 32000;
if (SampleRate>((22050+24000)/2)) return 24000;
if (SampleRate>((16000+22050)/2)) return 22050;
if (SampleRate>((12000+16000)/2)) return 16000;
if (SampleRate>((11025+12000)/2)) return 12000;
if (SampleRate>((8000+11025)/2)) return 11025;
return 8000;
}
void SB_ioport_out( WORD port, BYTE val )
{
switch(port)
{
/* DSP - Reset */
case 0x226:
TRACE("Resetting DSP.\n");
SB_Reset();
break;
/* DSP - Write Data or Command */
case 0x22c:
TRACE("val=%x\n",val);
if (command == -1) {
/* Clear input buffer and set the current command */
command = val;
InSize = 0;
}
if (InSize!=DSP_Command[command])
/* Fill the input buffer the command parameters if any */
DSP_InBuffer[InSize++]=val;
else {
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/* Process command */
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switch(command)
{
case 0x10: /* SB */
FIXME("Direct DAC (8-bit) - Not Implemented\n");
break;
case 0x14: /* SB */
SamplesCount = DSP_InBuffer[1]+(val<<8)+1;
TRACE("DMA DAC (8-bit) for %x samples\n",SamplesCount);
dma_enable = 1;
break;
case 0x20:
FIXME("Direct ADC (8-bit) - Not Implemented\n");
break;
case 0x24: /* SB */
FIXME("DMA ADC (8-bit) - Not Implemented\n");
break;
case 0x40: /* SB */
SampleRate = 1000000/(256-val);
TRACE("Set Time Constant (%d <-> %d Hz => %d Hz)\n",DSP_InBuffer[0],
SampleRate,SB_StdSampleRate(SampleRate));
SampleRate = SB_StdSampleRate(SampleRate);
wav_fmt.nSamplesPerSec = SampleRate;
wav_fmt.nAvgBytesPerSec = SampleRate;
IDirectSoundBuffer_SetFormat(lpdsbuf,&wav_fmt);
break;
/* case 0xBX/0xCX -> See below */
case 0xD0: /* SB */
TRACE("Halt DMA operation (8-bit)\n");
dma_enable = 0;
break;
case 0xD1: /* SB */
FIXME("Enable Speaker - Not Implemented\n");
break;
case 0xD3: /* SB */
FIXME("Disable Speaker - Not Implemented\n");
break;
case 0xD4: /* SB */
FIXME("Continue DMA operation (8-bit) - Not Implemented\n");
break;
case 0xD8: /* SB */
FIXME("Speaker Status - Not Implemented\n");
break;
case 0xE0: /* SB 2.0 */
TRACE("DSP Identification\n");
DSP_OutBuffer[OutSize++] = ~val;
break;
case 0xE1: /* SB */
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TRACE("DSP Version\n");
OutSize=2;
DSP_OutBuffer[0]=0; /* returns version 1.0 */
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");
DOSVM_QueueEvent(SB_IRQ,SB_IRQ_PRI,NULL,NULL);
break;
default:
if (((command&0xF0)==0xB0)||((DSP_InBuffer[0]&0xF0)==0xC0)) {
/* SB16 */
FIXME("Generic DAC/ADC DMA (16-bit, 8-bit) - %d % d\n",command,DSP_InBuffer[1]);
if (command&0x02)
FIXME("Generic DAC/ADC fifo mode not supported\n");
if (command&0x04)
FIXME("Generic DAC/ADC autoinit dma mode not supported\n");
if (command&0x08)
FIXME("Generic DAC/ADC adc mode not supported\n");
switch(command>>4) {
case 0xB:
FIXME("Generic DAC/ADC 8-bit not supported\n");
SampleMode = 0;
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break;
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case 0xC:
FIXME("Generic DAC/ADC 16-bit not supported\n");
SampleMode = 1;
break;
default:
ERR("Generic DAC/ADC resolution unknown\n");
break;
}
if (DSP_InBuffer[1]&0x010)
FIXME("Generic DAC/ADC signed sample mode not supported\n");
if (DSP_InBuffer[1]&0x020)
FIXME("Generic DAC/ADC stereo mode not supported\n");
SamplesCount = DSP_InBuffer[2]+(val<<8)+1;
TRACE("Generic DMA for %x samples\n",SamplesCount);
dma_enable = 1;
}
else
FIXME("DSP command %x not supported\n",val);
}
/* Empty the input buffer and end the command */
InSize = 0;
command = -1;
}
}
}
BYTE SB_ioport_in( WORD port )
{
BYTE res = 0;
switch(port)
{
/* DSP Read Data */
case 0x22a:
/* Value in the read buffer */
if (OutSize)
res = DSP_OutBuffer[--OutSize];
else
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/* return the last byte */
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res = DSP_OutBuffer[0];
break;
/* DSP - Write Buffer Status */
case 0x22c:
/* DSP always ready for writing */
res = 0x00;
break;
/* DSP - Data Available Status */
/* DSP - IRQ Acknowledge, 8-bit */
case 0x22e:
/* DSP data availability check */
if (OutSize)
res = 0x80;
else
res = 0x00;
break;
}
return res;
}