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-rw-r--r--plugins/ao/eng_ssf/eng_ssf.c316
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diff --git a/plugins/ao/eng_ssf/eng_ssf.c b/plugins/ao/eng_ssf/eng_ssf.c
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+++ b/plugins/ao/eng_ssf/eng_ssf.c
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+/*
+ 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 <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include "ao.h"
+#include "eng_protos.h"
+#include "corlett.h"
+#include "sat_hw.h"
+#include "scsp.h"
+
+#define DEBUG_LOADER (0)
+
+static corlett_t *c = NULL;
+static char psfby[256];
+static uint32 decaybegin, decayend, total_samples;
+
+void *scsp_start(const void *config);
+void SCSP_Update(void *param, INT16 **inputs, INT16 **buf, int samples);
+
+int32 ssf_start(uint8 *buffer, uint32 length)
+{
+ uint8 *file, *lib_decoded, *lib_raw_file;
+ uint32 offset, plength, lengthMS, fadeMS;
+ uint64 file_len, lib_len, lib_raw_length;
+ corlett_t *lib;
+ char *libfile;
+ int i;
+
+ // clear Saturn work RAM before we start scribbling in it
+ memset(sat_ram, 0, 512*1024);
+
+ // Decode the current SSF
+ if (corlett_decode(buffer, length, &file, &file_len, &c) != AO_SUCCESS)
+ {
+ return AO_FAIL;
+ }
+
+ #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 ? c->libaux[i-1] : c->lib;
+ if (libfile[0] != 0)
+ {
+ uint64 tmp_length;
+
+ #if DEBUG_LOADER
+ printf("Loading library: %s\n", c->lib);
+ #endif
+ if (ao_get_lib(libfile, &lib_raw_file, &tmp_length) != AO_SUCCESS)
+ {
+ return AO_FAIL;
+ }
+ 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);
+ return AO_FAIL;
+ }
+
+ // 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(&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(&sat_ram[offset], file+4, file_len-4);
+
+ free(file);
+
+ // Finally, set psfby tag
+ strcpy(psfby, "n/a");
+ if (c)
+ {
+ for (i = 0; i < MAX_UNKNOWN_TAGS; i++)
+ {
+ if (!strcasecmp(c->tag_name[i], "psfby"))
+ strcpy(psfby, c->tag_data[i]);
+ }
+ }
+
+ #if DEBUG_LOADER && 1
+ {
+ FILE *f;
+
+ f = fopen("satram.bin", "wb");
+ fwrite(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 = sat_ram[i];
+ sat_ram[i] = sat_ram[i+1];
+ sat_ram[i+1] = temp;
+ }
+
+ sat_hw_init();
+
+ // now figure out the time in samples for the length/fade
+ lengthMS = psfTimeToMS(c->inf_length);
+ fadeMS = psfTimeToMS(c->inf_fade);
+ total_samples = 0;
+
+ if (lengthMS == 0)
+ {
+ lengthMS = ~0;
+ }
+
+ if (lengthMS == ~0)
+ {
+ decaybegin = lengthMS;
+ }
+ else
+ {
+ lengthMS = (lengthMS * 441) / 10;
+ fadeMS = (fadeMS * 441) / 10;
+
+ decaybegin = lengthMS;
+ decayend = lengthMS + fadeMS;
+ }
+
+ return AO_SUCCESS;
+}
+
+int32 ssf_gen(int16 *buffer, uint32 samples)
+{
+ int i;
+ int16 output[samples], output2[samples];
+ int16 *stereo[2];
+ int16 *outp = buffer;
+
+ for (i = 0; i < samples; i++)
+ {
+ m68k_execute((11300000/60)/735);
+ stereo[0] = &output[i];
+ stereo[1] = &output2[i];
+ SCSP_Update(NULL, NULL, stereo, 1);
+ }
+
+ for (i = 0; i < samples; i++)
+ {
+ // process the fade tags
+ if (total_samples >= decaybegin)
+ {
+ if (total_samples >= 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*(total_samples - decaybegin)/(decayend-decaybegin));
+ output[i] = (output[i] * fader)>>8;
+ output2[i] = (output2[i] * fader)>>8;
+
+ total_samples++;
+ }
+ }
+ else
+ {
+ total_samples++;
+ }
+
+ *outp++ = output[i];
+ *outp++ = output2[i];
+ }
+
+ return AO_SUCCESS;
+}
+
+int32 ssf_stop(void)
+{
+ return AO_SUCCESS;
+}
+
+int32 ssf_command(int32 command, int32 parameter)
+
+{
+ switch (command)
+ {
+ case COMMAND_RESTART:
+ return AO_SUCCESS;
+
+ }
+ return AO_FAIL;
+}
+
+int32 ssf_fill_info(ao_display_info *info)
+{
+ if (c == NULL)
+ return AO_FAIL;
+
+ strcpy(info->title[1], "Name: ");
+ sprintf(info->info[1], "%s", c->inf_title);
+
+ strcpy(info->title[2], "Game: ");
+ sprintf(info->info[2], "%s", c->inf_game);
+
+ strcpy(info->title[3], "Artist: ");
+ sprintf(info->info[3], "%s", c->inf_artist);
+
+ strcpy(info->title[4], "Copyright: ");
+ sprintf(info->info[4], "%s", c->inf_copy);
+
+ strcpy(info->title[5], "Year: ");
+ sprintf(info->info[5], "%s", c->inf_year);
+
+ strcpy(info->title[6], "Length: ");
+ sprintf(info->info[6], "%s", c->inf_length);
+
+ strcpy(info->title[7], "Fade: ");
+ sprintf(info->info[7], "%s", c->inf_fade);
+
+ strcpy(info->title[8], "Ripper: ");
+ sprintf(info->info[8], "%s", psfby);
+
+ return AO_SUCCESS;
+}