Removed outdated Dreamcast support
--HG-- extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%403813
This commit is contained in:
Sam Lantinga
parent
6ecdaa1fe3
commit
232c747269
39 changed files with 8 additions and 2772 deletions
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@ -140,9 +140,6 @@ static const AudioBootStrap *const bootstrap[] = {
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#if SDL_AUDIO_DRIVER_DUMMY
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&DUMMYAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_DC
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&DCAUD_bootstrap,
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#endif
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#if SDL_AUDIO_DRIVER_MMEAUDIO
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&MMEAUDIO_bootstrap,
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#endif
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@ -1,245 +0,0 @@
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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2009 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 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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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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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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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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/* Output dreamcast aica */
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#include "SDL_timer.h"
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#include "SDL_audio.h"
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#include "../SDL_audiomem.h"
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#include "../SDL_audio_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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#define SPU_RAM_BASE 0xa0800000
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static void
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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
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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
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DCAUD_PlayDevice(_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,
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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,
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this->hidden->rightpos + offset,
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this->hidden->mixbuf, this->hidden->mixlen);
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} else {
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spu_memload_stereo16(this->hidden->leftpos + offset,
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this->hidden->rightpos + offset,
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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,
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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,
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spec->samples * 2, spec->freq, 255, 0, 1);
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aica_play(1, mode, this->hidden->rightpos, 0,
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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 *
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DCAUD_GetDeviceBuf(_THIS)
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{
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return (this->hidden->mixbuf);
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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
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DCAUD_WaitDevice(_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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static void
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DCAUD_CloseDevice(_THIS)
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{
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if (this->hidden != NULL) {
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aica_stop(0);
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if (this->spec.channels == 2) {
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aica_stop(1);
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}
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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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SDL_free(this->hidden);
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this->hidden = NULL;
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/* !!! FIXME: is there a reverse of spu_init()? */
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}
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}
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static int
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DCAUD_OpenDevice(_THIS, SDL_AudioSpec * spec)
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{
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SDL_AudioFormat test_format = SDL_FirstAudioFormat(spec->format);
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int valid_datatype = 0;
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/* Initialize all variables that we clean on shutdown */
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this->hidden = (struct SDL_PrivateAudioData *)
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SDL_malloc((sizeof *this->hidden));
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if (this->hidden == NULL) {
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SDL_OutOfMemory();
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return 0;
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}
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SDL_memset(this->hidden, 0, (sizeof *this->hidden));
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spu_init();
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while ((!valid_datatype) && (test_format)) {
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spec->format = test_format;
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switch (test_format) {
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/* only formats Dreamcast accepts... */
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case AUDIO_S8:
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case AUDIO_S16LSB:
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valid_datatype = 1;
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break;
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default:
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test_format = SDL_NextAudioFormat();
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break;
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}
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}
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if (!valid_datatype) { /* shouldn't happen, but just in case... */
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DCAUD_CloseDevice(this);
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SDL_SetError("Unsupported audio format");
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return 0;
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}
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if (spec->channels > 2)
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spec->channels = 2; /* no more than stereo on the Dreamcast. */
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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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DCAUD_CloseDevice(this);
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SDL_OutOfMemory();
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return 0;
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}
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SDL_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 1;
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}
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static int
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DCAUD_Init(SDL_AudioDriverImpl * impl)
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{
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/* Set the function pointers */
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impl->OpenDevice = DCAUD_OpenDevice;
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impl->PlayDevice = DCAUD_PlayDevice;
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impl->WaitDevice = DCAUD_WaitDevice;
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impl->GetDeviceBuf = DCAUD_GetDeviceBuf;
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impl->CloseDevice = DCAUD_CloseDevice;
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impl->OnlyHasDefaultOutputDevice = 1;
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return 1;
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}
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AudioBootStrap DCAUD_bootstrap = {
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"dcaudio", "Dreamcast AICA audio", DCAUD_Init, 0
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};
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/* vi: set ts=4 sw=4 expandtab: */
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@ -1,43 +0,0 @@
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/*
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SDL - Simple DirectMedia Layer
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Copyright (C) 1997-2009 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 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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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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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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Sam Lantinga
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slouken@libsdl.org
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*/
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#include "SDL_config.h"
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#ifndef _SDL_dcaudio_h
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#define _SDL_dcaudio_h
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#include "../SDL_sysaudio.h"
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/* Hidden "this" pointer for the audio functions */
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#define _THIS SDL_AudioDevice *this
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struct SDL_PrivateAudioData
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{
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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_dcaudio_h */
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/* vi: set ts=4 sw=4 expandtab: */
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@ -1,301 +0,0 @@
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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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#include <arch/irq.h>
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#include <dc/spu.h>
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#if 0
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#define dc_snd_base ((volatile unsigned char *)0x00800000) /* arm side */
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#endif
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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
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aica_init()
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{
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int i, j, old = 0;
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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)
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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,
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177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185,
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186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194,
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195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202,
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203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210,
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211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218,
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219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225,
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226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233,
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233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240,
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240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246,
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247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255
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};
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/* For the moment this is going to have to suffice, until we really
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figure out what these mean. */
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#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f)))
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#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)])
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//#define AICA_VOL(x) (0xff - logs[x&255])
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static inline unsigned
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AICA_FREQ(unsigned freq)
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{
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unsigned long freq_lo, freq_base = 5644800;
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int freq_hi = 7;
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/* Need to convert frequency to floating point format
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(freq_hi is exponent, freq_lo is mantissa)
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Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */
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while (freq < freq_base && freq_hi > -8) {
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freq_base >>= 1;
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--freq_hi;
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}
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while (freq < freq_base && freq_hi > -8) {
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freq_base >>= 1;
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freq_hi--;
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}
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freq_lo = (freq << 10) / freq_base;
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return (freq_hi << 11) | (freq_lo & 1023);
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}
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/* Sets up a sound channel completely. This is generally good if you want
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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 = 0;
|
||||
|
||||
/* 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
|
||||
}
|
||||
|
||||
/* vi: set ts=4 sw=4 expandtab: */
|
||||
|
|
@ -1,42 +0,0 @@
|
|||
/*
|
||||
SDL - Simple DirectMedia Layer
|
||||
Copyright (C) 1997-2009 Sam Lantinga
|
||||
|
||||
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., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
|
||||
|
||||
Sam Lantinga
|
||||
slouken@libsdl.org
|
||||
*/
|
||||
#include "SDL_config.h"
|
||||
|
||||
#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
|
||||
/* vi: set ts=4 sw=4 expandtab: */
|
||||
Loading…
Add table
Add a link
Reference in a new issue