diff options
Diffstat (limited to '3rdparty/SDL/src/audio/dc')
| -rw-r--r-- | 3rdparty/SDL/src/audio/dc/SDL_dcaudio.c | 246 | ||||
| -rw-r--r-- | 3rdparty/SDL/src/audio/dc/SDL_dcaudio.h | 41 | ||||
| -rw-r--r-- | 3rdparty/SDL/src/audio/dc/aica.c | 271 | ||||
| -rw-r--r-- | 3rdparty/SDL/src/audio/dc/aica.h | 40 |
4 files changed, 598 insertions, 0 deletions
diff --git a/3rdparty/SDL/src/audio/dc/SDL_dcaudio.c b/3rdparty/SDL/src/audio/dc/SDL_dcaudio.c new file mode 100644 index 0000000..88daa87 --- /dev/null +++ b/3rdparty/SDL/src/audio/dc/SDL_dcaudio.c @@ -0,0 +1,246 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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" + +/* Output dreamcast aica */ + +#include "SDL_timer.h" +#include "SDL_audio.h" +#include "../SDL_audiomem.h" +#include "../SDL_audio_c.h" +#include "../SDL_audiodev_c.h" +#include "SDL_dcaudio.h" + +#include "aica.h" +#include <dc/spu.h> + +/* Audio driver functions */ +static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec); +static void DCAUD_WaitAudio(_THIS); +static void DCAUD_PlayAudio(_THIS); +static Uint8 *DCAUD_GetAudioBuf(_THIS); +static void DCAUD_CloseAudio(_THIS); + +/* Audio driver bootstrap functions */ +static int DCAUD_Available(void) +{ + return 1; +} + +static void DCAUD_DeleteDevice(SDL_AudioDevice *device) +{ + SDL_free(device->hidden); + SDL_free(device); +} + +static SDL_AudioDevice *DCAUD_CreateDevice(int devindex) +{ + SDL_AudioDevice *this; + + /* Initialize all variables that we clean on shutdown */ + this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); + if ( this ) { + SDL_memset(this, 0, (sizeof *this)); + this->hidden = (struct SDL_PrivateAudioData *) + SDL_malloc((sizeof *this->hidden)); + } + if ( (this == NULL) || (this->hidden == NULL) ) { + SDL_OutOfMemory(); + if ( this ) { + SDL_free(this); + } + return(0); + } + SDL_memset(this->hidden, 0, (sizeof *this->hidden)); + + /* Set the function pointers */ + this->OpenAudio = DCAUD_OpenAudio; + this->WaitAudio = DCAUD_WaitAudio; + this->PlayAudio = DCAUD_PlayAudio; + this->GetAudioBuf = DCAUD_GetAudioBuf; + this->CloseAudio = DCAUD_CloseAudio; + + this->free = DCAUD_DeleteDevice; + + spu_init(); + + return this; +} + +AudioBootStrap DCAUD_bootstrap = { + "dcaudio", "Dreamcast AICA audio", + DCAUD_Available, DCAUD_CreateDevice +}; + +/* This function waits until it is possible to write a full sound buffer */ +static void DCAUD_WaitAudio(_THIS) +{ + if (this->hidden->playing) { + /* wait */ + while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) { + thd_pass(); + } + } +} + +#define SPU_RAM_BASE 0xa0800000 + +static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size) +{ + uint8 *src = src0; + uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); + uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); + size = (size+7)/8; + while(size--) { + unsigned lval,rval; + lval = *src++; + rval = *src++; + lval|= (*src++)<<8; + rval|= (*src++)<<8; + lval|= (*src++)<<16; + rval|= (*src++)<<16; + lval|= (*src++)<<24; + rval|= (*src++)<<24; + g2_write_32(left++,lval); + g2_write_32(right++,rval); + g2_fifo_wait(); + } +} + +static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size) +{ + uint16 *src = src0; + uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE); + uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE); + size = (size+7)/8; + while(size--) { + unsigned lval,rval; + lval = *src++; + rval = *src++; + lval|= (*src++)<<16; + rval|= (*src++)<<16; + g2_write_32(left++,lval); + g2_write_32(right++,rval); + g2_fifo_wait(); + } +} + +static void DCAUD_PlayAudio(_THIS) +{ + SDL_AudioSpec *spec = &this->spec; + unsigned int offset; + + if (this->hidden->playing) { + /* wait */ + while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) { + thd_pass(); + } + } + + offset = this->hidden->nextbuf*spec->size; + this->hidden->nextbuf^=1; + /* Write the audio data, checking for EAGAIN on broken audio drivers */ + if (spec->channels==1) { + spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } else { + offset/=2; + if ((this->spec.format&255)==8) { + spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } else { + spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen); + } + } + + if (!this->hidden->playing) { + int mode; + this->hidden->playing = 1; + mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT; + if (spec->channels==1) { + aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1); + } else { + aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1); + aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1); + } + } +} + +static Uint8 *DCAUD_GetAudioBuf(_THIS) +{ + return(this->hidden->mixbuf); +} + +static void DCAUD_CloseAudio(_THIS) +{ + aica_stop(0); + if (this->spec.channels==2) aica_stop(1); + if ( this->hidden->mixbuf != NULL ) { + SDL_FreeAudioMem(this->hidden->mixbuf); + this->hidden->mixbuf = NULL; + } +} + +static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec) +{ + Uint16 test_format = SDL_FirstAudioFormat(spec->format); + int valid_datatype = 0; + while ((!valid_datatype) && (test_format)) { + spec->format = test_format; + switch (test_format) { + /* only formats Dreamcast accepts... */ + case AUDIO_S8: + case AUDIO_S16LSB: + valid_datatype = 1; + break; + + default: + test_format = SDL_NextAudioFormat(); + break; + } + } + + if (!valid_datatype) { /* shouldn't happen, but just in case... */ + SDL_SetError("Unsupported audio format"); + return (-1); + } + + if (spec->channels > 2) + spec->channels = 2; /* no more than stereo on the Dreamcast. */ + + /* Update the fragment size as size in bytes */ + SDL_CalculateAudioSpec(spec); + + /* Allocate mixing buffer */ + this->hidden->mixlen = spec->size; + this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); + if ( this->hidden->mixbuf == NULL ) { + return(-1); + } + SDL_memset(this->hidden->mixbuf, spec->silence, spec->size); + this->hidden->leftpos = 0x11000; + this->hidden->rightpos = 0x11000+spec->size; + this->hidden->playing = 0; + this->hidden->nextbuf = 0; + + /* We're ready to rock and roll. :-) */ + return(0); +} diff --git a/3rdparty/SDL/src/audio/dc/SDL_dcaudio.h b/3rdparty/SDL/src/audio/dc/SDL_dcaudio.h new file mode 100644 index 0000000..fba95b3 --- /dev/null +++ b/3rdparty/SDL/src/audio/dc/SDL_dcaudio.h @@ -0,0 +1,41 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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 _SDL_dcaudio_h +#define _SDL_dcaudio_h + +#include "../SDL_sysaudio.h" + +/* Hidden "this" pointer for the video functions */ +#define _THIS SDL_AudioDevice *this + +struct SDL_PrivateAudioData { + /* The file descriptor for the audio device */ + Uint8 *mixbuf; + Uint32 mixlen; + int playing; + int leftpos,rightpos; + int nextbuf; +}; + +#endif /* _SDL_dcaudio_h */ diff --git a/3rdparty/SDL/src/audio/dc/aica.c b/3rdparty/SDL/src/audio/dc/aica.c new file mode 100644 index 0000000..b6a1c93 --- /dev/null +++ b/3rdparty/SDL/src/audio/dc/aica.c @@ -0,0 +1,271 @@ +/* This file is part of the Dreamcast function library. + * Please see libdream.c for further details. + * + * (c)2000 Dan Potter + * modify BERO + */ +#include "aica.h" + +#include <arch/irq.h> +#include <dc/spu.h> + +/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */ +#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */ + +/* Some convienence macros */ +#define SNDREGADDR(x) (0xa0700000 + (x)) +#define CHNREGADDR(ch,x) SNDREGADDR(0x80*(ch)+(x)) + + +#define SNDREG32(x) (*(volatile unsigned long *)SNDREGADDR(x)) +#define SNDREG8(x) (*(volatile unsigned char *)SNDREGADDR(x)) +#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) +#define CHNREG8(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x)) + +#define G2_LOCK(OLD) \ + do { \ + if (!irq_inside_int()) \ + OLD = irq_disable(); \ + /* suspend any G2 DMA here... */ \ + while((*(volatile unsigned int *)0xa05f688c) & 0x20) \ + ; \ + } while(0) + +#define G2_UNLOCK(OLD) \ + do { \ + /* resume any G2 DMA here... */ \ + if (!irq_inside_int()) \ + irq_restore(OLD); \ + } while(0) + + +void aica_init() { + int i, j, old = 0; + + /* Initialize AICA channels */ + G2_LOCK(old); + SNDREG32(0x2800) = 0x0000; + + for (i=0; i<64; i++) { + for (j=0; j<0x80; j+=4) { + if ((j&31)==0) g2_fifo_wait(); + CHNREG32(i, j) = 0; + } + g2_fifo_wait(); + CHNREG32(i,0) = 0x8000; + CHNREG32(i,20) = 0x1f; + } + + SNDREG32(0x2800) = 0x000f; + g2_fifo_wait(); + G2_UNLOCK(old); +} + +/* Translates a volume from linear form to logarithmic form (required by + the AICA chip */ +/* int logs[] = { + +0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103, +105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127, +129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145, +146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159, +160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171, +172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182, +182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191, +191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199, +200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207, +208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215, +215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222, +222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228, +228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234, +234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240, +240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245, +246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251, +251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255 + +}; */ + +const static unsigned char logs[] = { + 0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61, + 63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88, + 90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106, + 108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121, + 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134, + 135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146, + 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156, + 157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167, + 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 = 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 +} diff --git a/3rdparty/SDL/src/audio/dc/aica.h b/3rdparty/SDL/src/audio/dc/aica.h new file mode 100644 index 0000000..2721e42 --- /dev/null +++ b/3rdparty/SDL/src/audio/dc/aica.h @@ -0,0 +1,40 @@ +/* + SDL - Simple DirectMedia Layer + Copyright (C) 1997-2012 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 |