Added initial support for Dreamcast (thanks HERO!)
--HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%40510
This commit is contained in:
Sam Lantinga
parent
148a1a64f4
commit
3ee2de057f
43 changed files with 2847 additions and 4 deletions
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@ -6,7 +6,7 @@ noinst_LTLIBRARIES = libaudio.la
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# Define which subdirectories need to be built
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SUBDIRS = @AUDIO_SUBDIRS@
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DIST_SUBDIRS = alsa arts baudio dma dmedia dsp esd macrom nas nto openbsd \
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paudio sun ums windib windx5 disk mint
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paudio sun ums windib windx5 disk mint dc
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DRIVERS = @AUDIO_DRIVERS@
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@ -88,6 +88,9 @@ static AudioBootStrap *bootstrap[] = {
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#endif
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#ifdef DISKAUD_SUPPORT
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&DISKAUD_bootstrap,
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#endif
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#ifdef ENABLE_DC
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&DCAUD_bootstrap,
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#endif
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NULL
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};
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@ -153,6 +153,9 @@ extern AudioBootStrap MINTAUDIO_bootstrap;
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#ifdef DISKAUD_SUPPORT
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extern AudioBootStrap DISKAUD_bootstrap;
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#endif
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#ifdef ENABLE_DC
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extern AudioBootStrap DCAUD_bootstrap;
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#endif
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/* This is the current audio device */
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extern SDL_AudioDevice *current_audio;
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6
src/audio/dc/.cvsignore
Normal file
6
src/audio/dc/.cvsignore
Normal file
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@ -0,0 +1,6 @@
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Makefile.in
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Makefile
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.libs
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*.o
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*.lo
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*.la
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11
src/audio/dc/Makefile.am
Normal file
11
src/audio/dc/Makefile.am
Normal file
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@ -0,0 +1,11 @@
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## Makefile.am for SDL on the Dreamcast console
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noinst_LTLIBRARIES = libaudio_dc.la
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libaudio_dc_la_SOURCES = $(SRCS)
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# The SDL audio driver sources
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SRCS = SDL_dcaudio.c \
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SDL_dcaudio.h \
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aica.c \
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aica.h
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245
src/audio/dc/SDL_dcaudio.c
Normal file
245
src/audio/dc/SDL_dcaudio.c
Normal file
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@ -0,0 +1,245 @@
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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002 Sam Lantinga
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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 Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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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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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public
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License along with this library; if not, write to the Free
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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BERO <bero@geocities.co.jp>
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based on SDL_diskaudio.c by Sam Lantinga <slouken@libsdl.org>
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*/
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#ifdef SAVE_RCSID
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static char rcsid =
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"@(#) $Id$";
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#endif
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/* Output dreamcast aica */
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <errno.h>
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#include <unistd.h>
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#include <sys/stat.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include "SDL_audio.h"
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#include "SDL_error.h"
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#include "SDL_audiomem.h"
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#include "SDL_audio_c.h"
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#include "SDL_timer.h"
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#include "SDL_audiodev_c.h"
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#include "SDL_dcaudio.h"
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#include "aica.h"
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#include <dc/spu.h>
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/* Audio driver functions */
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static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
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static void DCAUD_WaitAudio(_THIS);
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static void DCAUD_PlayAudio(_THIS);
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static Uint8 *DCAUD_GetAudioBuf(_THIS);
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static void DCAUD_CloseAudio(_THIS);
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/* Audio driver bootstrap functions */
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static int DCAUD_Available(void)
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{
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return 1;
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}
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static void DCAUD_DeleteDevice(SDL_AudioDevice *device)
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{
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free(device->hidden);
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free(device);
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}
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static SDL_AudioDevice *DCAUD_CreateDevice(int devindex)
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{
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SDL_AudioDevice *this;
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/* Initialize all variables that we clean on shutdown */
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this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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if ( this ) {
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memset(this, 0, (sizeof *this));
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this->hidden = (struct SDL_PrivateAudioData *)
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malloc((sizeof *this->hidden));
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}
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if ( (this == NULL) || (this->hidden == NULL) ) {
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SDL_OutOfMemory();
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if ( this ) {
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free(this);
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}
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return(0);
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}
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memset(this->hidden, 0, (sizeof *this->hidden));
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/* Set the function pointers */
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this->OpenAudio = DCAUD_OpenAudio;
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this->WaitAudio = DCAUD_WaitAudio;
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this->PlayAudio = DCAUD_PlayAudio;
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this->GetAudioBuf = DCAUD_GetAudioBuf;
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this->CloseAudio = DCAUD_CloseAudio;
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this->free = DCAUD_DeleteDevice;
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spu_init();
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return this;
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}
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AudioBootStrap DCAUD_bootstrap = {
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"dcaudio", "Dreamcast AICA audio",
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DCAUD_Available, DCAUD_CreateDevice
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};
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/* This function waits until it is possible to write a full sound buffer */
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static void DCAUD_WaitAudio(_THIS)
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{
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if (this->hidden->playing) {
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/* wait */
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while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) {
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thd_pass();
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}
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}
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}
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#define SPU_RAM_BASE 0xa0800000
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static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size)
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{
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uint8 *src = src0;
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uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE);
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uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
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size = (size+7)/8;
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while(size--) {
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unsigned lval,rval;
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lval = *src++;
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rval = *src++;
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lval|= (*src++)<<8;
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rval|= (*src++)<<8;
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lval|= (*src++)<<16;
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rval|= (*src++)<<16;
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lval|= (*src++)<<24;
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rval|= (*src++)<<24;
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g2_write_32(left++,lval);
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g2_write_32(right++,rval);
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g2_fifo_wait();
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}
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}
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static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size)
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{
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uint16 *src = src0;
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uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE);
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uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
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size = (size+7)/8;
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while(size--) {
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unsigned lval,rval;
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lval = *src++;
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rval = *src++;
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lval|= (*src++)<<16;
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rval|= (*src++)<<16;
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g2_write_32(left++,lval);
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g2_write_32(right++,rval);
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g2_fifo_wait();
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}
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}
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static void DCAUD_PlayAudio(_THIS)
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{
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SDL_AudioSpec *spec = &this->spec;
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unsigned int offset;
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if (this->hidden->playing) {
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/* wait */
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while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) {
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thd_pass();
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}
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}
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offset = this->hidden->nextbuf*spec->size;
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this->hidden->nextbuf^=1;
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/* Write the audio data, checking for EAGAIN on broken audio drivers */
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if (spec->channels==1) {
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spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
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} else {
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offset/=2;
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if ((this->spec.format&255)==8) {
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spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
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} else {
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spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
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}
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}
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if (!this->hidden->playing) {
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int mode;
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this->hidden->playing = 1;
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mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT;
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if (spec->channels==1) {
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aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1);
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} else {
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aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1);
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aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1);
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}
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}
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}
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static Uint8 *DCAUD_GetAudioBuf(_THIS)
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{
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return(this->hidden->mixbuf);
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}
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static void DCAUD_CloseAudio(_THIS)
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{
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aica_stop(0);
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if (this->spec.channels==2) aica_stop(1);
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if ( this->hidden->mixbuf != NULL ) {
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SDL_FreeAudioMem(this->hidden->mixbuf);
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this->hidden->mixbuf = NULL;
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}
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}
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static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
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{
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switch(spec->format&0xff) {
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case 8: spec->format = AUDIO_S8; break;
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case 16: spec->format = AUDIO_S16LSB; break;
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default:
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SDL_SetError("Unsupported audio format");
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return(-1);
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}
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/* Update the fragment size as size in bytes */
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SDL_CalculateAudioSpec(spec);
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/* Allocate mixing buffer */
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this->hidden->mixlen = spec->size;
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this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
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if ( this->hidden->mixbuf == NULL ) {
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return(-1);
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}
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memset(this->hidden->mixbuf, spec->silence, spec->size);
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this->hidden->leftpos = 0x11000;
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this->hidden->rightpos = 0x11000+spec->size;
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this->hidden->playing = 0;
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this->hidden->nextbuf = 0;
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/* We're ready to rock and roll. :-) */
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return(0);
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}
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45
src/audio/dc/SDL_dcaudio.h
Normal file
45
src/audio/dc/SDL_dcaudio.h
Normal file
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@ -0,0 +1,45 @@
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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002 Sam Lantinga
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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 Library General Public
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License as published by the Free Software Foundation; either
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version 2 of the License, or (at your option) any later version.
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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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Library General Public License for more details.
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You should have received a copy of the GNU Library General Public
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License along with this library; if not, write to the Free
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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BERO <bero@geocities.co.jp>
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based on SDL_diskaudio.h by Sam Lantinga <slouken@libsdl.org>
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*/
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#ifdef SAVE_RCSID
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static char rcsid =
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"@(#) $Id$";
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#endif
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#ifndef _SDL_diskaudio_h
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#define _SDL_diskaudio_h
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#include "SDL_sysaudio.h"
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/* Hidden "this" pointer for the video functions */
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#define _THIS SDL_AudioDevice *this
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struct SDL_PrivateAudioData {
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/* The file descriptor for the audio device */
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Uint8 *mixbuf;
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Uint32 mixlen;
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int playing;
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int leftpos,rightpos;
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int nextbuf;
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};
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#endif /* _SDL_diskaudio_h */
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267
src/audio/dc/aica.c
Normal file
267
src/audio/dc/aica.c
Normal file
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@ -0,0 +1,267 @@
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/* This file is part of the Dreamcast function library.
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* Please see libdream.c for further details.
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*
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* (c)2000 Dan Potter
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* modify BERO
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*/
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#include "aica.h"
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/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */
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#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */
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/* Some convienence macros */
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#define SNDREGADDR(x) (0xa0700000 + (x))
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#define CHNREGADDR(ch,x) SNDREGADDR(0x80*(ch)+(x))
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#define SNDREG32(x) (*(volatile unsigned long *)SNDREGADDR(x))
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#define SNDREG8(x) (*(volatile unsigned char *)SNDREGADDR(x))
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#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
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#define CHNREG8(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
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#define G2_LOCK(OLD) \
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do { \
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if (!irq_inside_int()) \
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OLD = irq_disable(); \
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/* suspend any G2 DMA here... */ \
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while((*(volatile unsigned int *)0xa05f688c) & 0x20) \
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; \
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} while(0)
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#define G2_UNLOCK(OLD) \
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do { \
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/* resume any G2 DMA here... */ \
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if (!irq_inside_int()) \
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irq_restore(OLD); \
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} while(0)
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void aica_init() {
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int i, j, old;
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/* Initialize AICA channels */
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G2_LOCK(old);
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SNDREG32(0x2800) = 0x0000;
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for (i=0; i<64; i++) {
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for (j=0; j<0x80; j+=4) {
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if ((j&31)==0) g2_fifo_wait();
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CHNREG32(i, j) = 0;
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}
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g2_fifo_wait();
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CHNREG32(i,0) = 0x8000;
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CHNREG32(i,20) = 0x1f;
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}
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SNDREG32(0x2800) = 0x000f;
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g2_fifo_wait();
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G2_UNLOCK(old);
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}
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/* Translates a volume from linear form to logarithmic form (required by
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the AICA chip */
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/* int logs[] = {
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0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103,
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105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127,
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129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145,
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146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159,
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160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171,
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172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182,
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182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191,
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191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199,
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200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207,
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208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215,
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215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222,
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222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228,
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228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234,
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234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240,
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240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245,
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246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251,
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251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255
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}; */
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const static unsigned char logs[] = {
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0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61,
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63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88,
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90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106,
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108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121,
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122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134,
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135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146,
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146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156,
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157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167,
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167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176,
|
||||
177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185,
|
||||
186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194,
|
||||
195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202,
|
||||
203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210,
|
||||
211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218,
|
||||
219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225,
|
||||
226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233,
|
||||
233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240,
|
||||
240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246,
|
||||
247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255
|
||||
};
|
||||
|
||||
/* For the moment this is going to have to suffice, until we really
|
||||
figure out what these mean. */
|
||||
#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f)))
|
||||
#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)])
|
||||
//#define AICA_VOL(x) (0xff - logs[x&255])
|
||||
|
||||
static inline unsigned AICA_FREQ(unsigned freq) {
|
||||
unsigned long freq_lo, freq_base = 5644800;
|
||||
int freq_hi = 7;
|
||||
|
||||
/* Need to convert frequency to floating point format
|
||||
(freq_hi is exponent, freq_lo is mantissa)
|
||||
Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */
|
||||
while (freq < freq_base && freq_hi > -8) {
|
||||
freq_base >>= 1;
|
||||
--freq_hi;
|
||||
}
|
||||
while (freq < freq_base && freq_hi > -8) {
|
||||
freq_base >>= 1;
|
||||
freq_hi--;
|
||||
}
|
||||
freq_lo = (freq<<10) / freq_base;
|
||||
return (freq_hi << 11) | (freq_lo & 1023);
|
||||
}
|
||||
|
||||
/* Sets up a sound channel completely. This is generally good if you want
|
||||
a quick and dirty way to play notes. If you want a more comprehensive
|
||||
set of routines (more like PC wavetable cards) see below.
|
||||
|
||||
ch is the channel to play on (0 - 63)
|
||||
smpptr is the pointer to the sound data; if you're running off the
|
||||
SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just
|
||||
ptr. Basically, it's an offset into sound ram.
|
||||
mode is one of the mode constants (16 bit, 8 bit, ADPCM)
|
||||
nsamp is the number of samples to play (not number of bytes!)
|
||||
freq is the sampling rate of the sound
|
||||
vol is the volume, 0 to 0xff (0xff is louder)
|
||||
pan is a panning constant -- 0 is left, 128 is center, 255 is right.
|
||||
|
||||
This routine (and the similar ones) owe a lot to Marcus' sound example --
|
||||
I hadn't gotten quite this far into dissecting the individual regs yet. */
|
||||
void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) {
|
||||
int i;
|
||||
int val;
|
||||
int old;
|
||||
|
||||
/* Stop the channel (if it's already playing) */
|
||||
aica_stop(ch);
|
||||
/* doesn't seem to be needed, but it's here just in case */
|
||||
/*
|
||||
for (i=0; i<256; i++) {
|
||||
asm("nop");
|
||||
asm("nop");
|
||||
asm("nop");
|
||||
asm("nop");
|
||||
}
|
||||
*/
|
||||
G2_LOCK(old);
|
||||
/* Envelope setup. The first of these is the loop point,
|
||||
e.g., where the sample starts over when it loops. The second
|
||||
is the loop end. This is the full length of the sample when
|
||||
you are not looping, or the loop end point when you are (though
|
||||
storing more than that is a waste of memory if you're not doing
|
||||
volume enveloping). */
|
||||
CHNREG32(ch, 8) = loopst & 0xffff;
|
||||
CHNREG32(ch, 12) = loopend & 0xffff;
|
||||
|
||||
/* Write resulting values */
|
||||
CHNREG32(ch, 24) = AICA_FREQ(freq);
|
||||
|
||||
/* Set volume, pan, and some other things that we don't know what
|
||||
they do =) */
|
||||
CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8);
|
||||
/* Convert the incoming volume and pan into hardware values */
|
||||
/* Vol starts at zero so we can ramp */
|
||||
vol = AICA_VOL(vol);
|
||||
CHNREG32(ch, 40) = 0x24 | (vol<<8);
|
||||
/* Convert the incoming volume and pan into hardware values */
|
||||
/* Vol starts at zero so we can ramp */
|
||||
|
||||
/* If we supported volume envelopes (which we don't yet) then
|
||||
this value would set that up. The top 4 bits determine the
|
||||
envelope speed. f is the fastest, 1 is the slowest, and 0
|
||||
seems to be an invalid value and does weird things). The
|
||||
default (below) sets it into normal mode (play and terminate/loop).
|
||||
CHNREG32(ch, 16) = 0xf010;
|
||||
*/
|
||||
CHNREG32(ch, 16) = 0x1f; /* No volume envelope */
|
||||
|
||||
|
||||
/* Set sample format, buffer address, and looping control. If
|
||||
0x0200 mask is set on reg 0, the sample loops infinitely. If
|
||||
it's not set, the sample plays once and terminates. We'll
|
||||
also set the bits to start playback here. */
|
||||
CHNREG32(ch, 4) = smpptr & 0xffff;
|
||||
val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16);
|
||||
if (loopflag) val|=0x200;
|
||||
|
||||
CHNREG32(ch, 0) = val;
|
||||
|
||||
G2_UNLOCK(old);
|
||||
|
||||
/* Enable playback */
|
||||
/* CHNREG32(ch, 0) |= 0xc000; */
|
||||
g2_fifo_wait();
|
||||
|
||||
#if 0
|
||||
for (i=0xff; i>=vol; i--) {
|
||||
if ((i&7)==0) g2_fifo_wait();
|
||||
CHNREG32(ch, 40) = 0x24 | (i<<8);;
|
||||
}
|
||||
|
||||
g2_fifo_wait();
|
||||
#endif
|
||||
}
|
||||
|
||||
/* Stop the sound on a given channel */
|
||||
void aica_stop(int ch) {
|
||||
g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000);
|
||||
g2_fifo_wait();
|
||||
}
|
||||
|
||||
|
||||
/* The rest of these routines can change the channel in mid-stride so you
|
||||
can do things like vibrato and panning effects. */
|
||||
|
||||
/* Set channel volume */
|
||||
void aica_vol(int ch,int vol) {
|
||||
// g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol));
|
||||
g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) );
|
||||
g2_fifo_wait();
|
||||
}
|
||||
|
||||
/* Set channel pan */
|
||||
void aica_pan(int ch,int pan) {
|
||||
// g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan));
|
||||
g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) );
|
||||
g2_fifo_wait();
|
||||
}
|
||||
|
||||
/* Set channel frequency */
|
||||
void aica_freq(int ch,int freq) {
|
||||
g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq));
|
||||
g2_fifo_wait();
|
||||
}
|
||||
|
||||
/* Get channel position */
|
||||
int aica_get_pos(int ch) {
|
||||
#if 1
|
||||
/* Observe channel ch */
|
||||
g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8));
|
||||
g2_fifo_wait();
|
||||
/* Update position counters */
|
||||
return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
|
||||
#else
|
||||
/* Observe channel ch */
|
||||
g2_write_8(SNDREGADDR(0x280d),ch);
|
||||
/* Update position counters */
|
||||
return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
|
||||
#endif
|
||||
}
|
||||
17
src/audio/dc/aica.h
Normal file
17
src/audio/dc/aica.h
Normal file
|
|
@ -0,0 +1,17 @@
|
|||
#ifndef _AICA_H_
|
||||
#define _AICA_H_
|
||||
|
||||
#define AICA_MEM 0xa0800000
|
||||
|
||||
#define SM_8BIT 1
|
||||
#define SM_16BIT 0
|
||||
#define SM_ADPCM 2
|
||||
|
||||
void aica_play(int ch,int mode,unsigned long smpptr,int looptst,int loopend,int freq,int vol,int pan,int loopflag);
|
||||
void aica_stop(int ch);
|
||||
void aica_vol(int ch,int vol);
|
||||
void aica_pan(int ch,int pan);
|
||||
void aica_freq(int ch,int freq);
|
||||
int aica_get_pos(int ch);
|
||||
|
||||
#endif
|
||||
Loading…
Add table
Add a link
Reference in a new issue