/* Audio Overload SDK - SSF file format engine Copyright (c) 2007 R. Belmont and Richard Bannister. All rights reserved. Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: * Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. * Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. * Neither the names of R. Belmont and Richard Bannister nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ /* Sega driver commands: 00 - NOP 01 - SEQUENCE_START 02 - SEQUENCE_STOP 03 - SEQUENCE_PAUSE 04 - SEQUENCE_CONTINUE 05 - SEQUENCE_VOLUME 06 - SEQUENCE_ALLSTOP 07 - SEQUENCE_TEMPO 08 - SEQUENCE_MAP 09 - HOST_MIDI 0A - VOLUME_ANALYZE_START 0B - VOLUME_ANALYZE_STOP 0C - DSP CLEAR 0D - ALL OFF 0E - SEQUENCE PAN 0F - N/A 10 - SOUND INITIALIZE 11 - Yamaha 3D check (8C) 12 - QSound check (8B) 13 - Yamaha 3D init (8D) 80 - CD level 81 - CD pan 82 - MASTER VOLUME 83 - EFFECT_CHANGE 84 - NOP 85 - PCM stream play start 86 - PCM stream play end 87 - MIXER_CHANGE 88 - Mixer parameter change 89 - Hardware check 8A - PCM parameter change 8B - QSound check 8C - Yamaha 3D check 8D - Yamaha 3D init */ #include #include #include #include "ao.h" #include "eng_protos.h" #include "corlett.h" #include "sat_hw.h" #include "scsp.h" #include "m68kcpu.h" #define DEBUG_LOADER (0) typedef struct { corlett_t *c; char psfby[256]; uint32 decaybegin, decayend, total_samples; uint8 init_sat_ram[512*1024]; m68ki_cpu_core *cpu; } ssf_synth_t; void *ssf_start(const char *path, uint8 *buffer, uint32 length) { ssf_synth_t *s = malloc (sizeof (ssf_synth_t)); memset (s, 0, sizeof (ssf_synth_t)); uint8 *file, *lib_decoded, *lib_raw_file; uint32 offset, lengthMS, fadeMS; uint64 file_len, lib_len, lib_raw_length; corlett_t *lib; char *libfile; int i; s->cpu = m68k_init (); // // clear Saturn work RAM before we start scribbling in it // memset(cpu->sat_ram, 0, 512*1024); // Decode the current SSF if (corlett_decode(buffer, length, &file, &file_len, &s->c) != AO_SUCCESS) { ssf_stop (s); return NULL; } #if DEBUG_LOADER printf("%d bytes decoded\n", file_len); #endif // Get the library file, if any for (i=0; i<9; i++) { libfile = i ? s->c->libaux[i-1] : s->c->lib; if (libfile[0] != 0) { uint64 tmp_length; char libpath[PATH_MAX]; ao_getlibpath (path, s->c->lib, libpath, sizeof (libpath)); #if DEBUG_LOADER printf("Loading library: %s\n", c->lib); #endif if (ao_get_lib(libfile, &lib_raw_file, &tmp_length) != AO_SUCCESS) { ssf_stop (s); return NULL; } lib_raw_length = tmp_length; if (corlett_decode(lib_raw_file, lib_raw_length, &lib_decoded, &lib_len, &lib) != AO_SUCCESS) { free(lib_raw_file); ssf_stop (s); return NULL; } // Free up raw file free(lib_raw_file); // patch the file into ram offset = lib_decoded[0] | lib_decoded[1]<<8 | lib_decoded[2]<<16 | lib_decoded[3]<<24; // guard against invalid data if ((offset + (lib_len-4)) > 0x7ffff) { lib_len = 0x80000-offset+4; } memcpy(&s->cpu->sat_ram[offset], lib_decoded+4, lib_len-4); // Dispose the corlett structure for the lib - we don't use it free(lib); } } // now patch the file into RAM over the libraries offset = file[3]<<24 | file[2]<<16 | file[1]<<8 | file[0]; // guard against invalid data if ((offset + (file_len-4)) > 0x7ffff) { file_len = 0x80000-offset+4; } memcpy(&s->cpu->sat_ram[offset], file+4, file_len-4); free(file); // Finally, set psfby tag strcpy(s->psfby, "n/a"); if (s->c) { for (i = 0; i < MAX_UNKNOWN_TAGS; i++) { if (!strcasecmp(s->c->tag_name[i], "psfby")) strcpy(s->psfby, s->c->tag_data[i]); } } #if DEBUG_LOADER { FILE *f; f = fopen("satram.bin", "wb"); fwrite(cpu->sat_ram, 512*1024, 1, f); fclose(f); } #endif // now flip everything (this makes sense because he's using starscream) for (i = 0; i < 512*1024; i+=2) { uint8 temp; temp = s->cpu->sat_ram[i]; s->cpu->sat_ram[i] = s->cpu->sat_ram[i+1]; s->cpu->sat_ram[i+1] = temp; } // backup for fast restarting memcpy (s->init_sat_ram, s->cpu->sat_ram, sizeof (s->init_sat_ram)); sat_hw_init(s->cpu); // now figure out the time in samples for the length/fade lengthMS = psfTimeToMS(s->c->inf_length); fadeMS = psfTimeToMS(s->c->inf_fade); s->total_samples = 0; if (lengthMS == 0) { lengthMS = ~0; } if (lengthMS == ~0) { s->decaybegin = lengthMS; } else { lengthMS = (lengthMS * 441) / 10; fadeMS = (fadeMS * 441) / 10; s->decaybegin = lengthMS; s->decayend = lengthMS + fadeMS; } return s; } int32 ssf_gen(void *handle, int16 *buffer, uint32 samples) { ssf_synth_t *s = handle; int i; int16 output[samples], output2[samples]; int16 *stereo[2]; int16 *outp = buffer; for (i = 0; i < samples; i++) { m68k_execute(s->cpu, (11300000/60)/735); stereo[0] = &output[i]; stereo[1] = &output2[i]; SCSP_Update(s->cpu->SCSP, NULL, stereo, 1); } for (i = 0; i < samples; i++) { // process the fade tags if (s->total_samples >= s->decaybegin) { if (s->total_samples >= s->decayend) { // song is done here, call out as necessary to make your player stop output[i] = 0; output2[i] = 0; } else { int32 fader = 256 - (256*(s->total_samples - s->decaybegin)/(s->decayend-s->decaybegin)); output[i] = (output[i] * fader)>>8; output2[i] = (output2[i] * fader)>>8; s->total_samples++; } } else { s->total_samples++; } *outp++ = output[i]; *outp++ = output2[i]; } return AO_SUCCESS; } int32 ssf_stop(void *handle) { ssf_synth_t *s = handle; if (s) { if (s->c) { free (s->c); } if (s->cpu) { sat_hw_free (s->cpu); free (s->cpu); } free (s); } return AO_SUCCESS; } int32 ssf_command(void *handle, int32 command, int32 parameter) { ssf_synth_t *s = handle; switch (command) { case COMMAND_RESTART: sat_hw_free (s->cpu); memcpy (s->cpu->sat_ram, s->init_sat_ram, sizeof (s->init_sat_ram)); sat_hw_init (s->cpu); s->total_samples = 0; return AO_SUCCESS; } return AO_FAIL; } int32 ssf_fill_info(void *handle, ao_display_info *info) { ssf_synth_t *s = handle; if (s->c == NULL) return AO_FAIL; strcpy(info->title[1], "Name: "); sprintf(info->info[1], "%s", s->c->inf_title); strcpy(info->title[2], "Game: "); sprintf(info->info[2], "%s", s->c->inf_game); strcpy(info->title[3], "Artist: "); sprintf(info->info[3], "%s", s->c->inf_artist); strcpy(info->title[4], "Copyright: "); sprintf(info->info[4], "%s", s->c->inf_copy); strcpy(info->title[5], "Year: "); sprintf(info->info[5], "%s", s->c->inf_year); strcpy(info->title[6], "Length: "); sprintf(info->info[6], "%s", s->c->inf_length); strcpy(info->title[7], "Fade: "); sprintf(info->info[7], "%s", s->c->inf_fade); strcpy(info->title[8], "Ripper: "); sprintf(info->info[8], "%s", s->psfby); return AO_SUCCESS; }