diff options
Diffstat (limited to 'apps/codecs/spc/spc_dsp.c')
| -rw-r--r-- | apps/codecs/spc/spc_dsp.c | 1276 |
1 files changed, 0 insertions, 1276 deletions
diff --git a/apps/codecs/spc/spc_dsp.c b/apps/codecs/spc/spc_dsp.c deleted file mode 100644 index 153950c..0000000 --- a/apps/codecs/spc/spc_dsp.c +++ /dev/null @@ -1,1276 +0,0 @@ -/*************************************************************************** - * __________ __ ___. - * Open \______ \ ____ ____ | | _\_ |__ _______ ___ - * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / - * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < - * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ - * \/ \/ \/ \/ \/ - * $Id$ - * - * Copyright (C) 2007-2008 Michael Sevakis (jhMikeS) - * Copyright (C) 2006-2007 Adam Gashlin (hcs) - * Copyright (C) 2004-2007 Shay Green (blargg) - * Copyright (C) 2002 Brad Martin - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * as published by the Free Software Foundation; either version 2 - * of the License, or (at your option) any later version. - * - * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY - * KIND, either express or implied. - * - ****************************************************************************/ - -/* The DSP portion (awe!) */ -#include "codec.h" -#include "codecs.h" -#include "spc_codec.h" -#include "spc_profiler.h" - -#if defined(CPU_COLDFIRE) || defined (CPU_ARM) -int32_t fir_buf[FIR_BUF_CNT] - __attribute__ ((aligned (FIR_BUF_ALIGN*1))) IBSS_ATTR; -#endif -#if SPC_BRRCACHE -/* a little extra for samples that go past end */ -int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR; -#endif - -void DSP_write( struct Spc_Dsp* this, int i, int data ) -{ - assert( (unsigned) i < REGISTER_COUNT ); - - this->r.reg [i] = data; - int high = i >> 4; - int low = i & 0x0F; - if ( low < 2 ) /* voice volumes */ - { - int left = *(int8_t const*) &this->r.reg [i & ~1]; - int right = *(int8_t const*) &this->r.reg [i | 1]; - struct voice_t* v = this->voice_state + high; - v->volume [0] = left; - v->volume [1] = right; - } - else if ( low == 0x0F ) /* fir coefficients */ - { - this->fir_coeff [7 - high] = (int8_t) data; /* sign-extend */ - } -} - -/* if ( n < -32768 ) out = -32768; */ -/* if ( n > 32767 ) out = 32767; */ -#define CLAMP16( n ) \ -({ \ - if ( (int16_t) n != n ) \ - n = 0x7FFF ^ (n >> 31); \ - n; \ -}) - -#if SPC_BRRCACHE -static void decode_brr( struct Spc_Dsp* this, unsigned start_addr, - struct voice_t* voice, - struct raw_voice_t const* const raw_voice ) ICODE_ATTR; -static void decode_brr( struct Spc_Dsp* this, unsigned start_addr, - struct voice_t* voice, - struct raw_voice_t const* const raw_voice ) -{ - /* setup same variables as where decode_brr() is called from */ - #undef RAM - #define RAM ram.ram - struct src_dir const* const sd = - (struct src_dir*) &RAM [this->r.g.wave_page * 0x100]; - struct cache_entry_t* const wave_entry = - &this->wave_entry [raw_voice->waveform]; - - /* the following block can be put in place of the call to - decode_brr() below - */ - { - DEBUGF( "decode at %08x (wave #%d)\n", - start_addr, raw_voice->waveform ); - - /* see if in cache */ - int i; - for ( i = 0; i < this->oldsize; i++ ) - { - struct cache_entry_t* e = &this->wave_entry_old [i]; - if ( e->start_addr == start_addr ) - { - DEBUGF( "found in wave_entry_old (oldsize=%d)\n", - this->oldsize ); - *wave_entry = *e; - goto wave_in_cache; - } - } - - wave_entry->start_addr = start_addr; - - uint8_t const* const loop_ptr = - RAM + GET_LE16A( sd [raw_voice->waveform].loop ); - short* loop_start = 0; - - short* out = BRRcache + start_addr * 2; - wave_entry->samples = out; - *out++ = 0; - int smp1 = 0; - int smp2 = 0; - - uint8_t const* addr = RAM + start_addr; - int block_header; - do - { - if ( addr == loop_ptr ) - { - loop_start = out; - DEBUGF( "loop at %08lx (wave #%d)\n", - (unsigned long)(addr - RAM), raw_voice->waveform ); - } - - /* header */ - block_header = *addr; - addr += 9; - voice->addr = addr; - int const filter = (block_header & 0x0C) - 0x08; - - /* scaling - (invalid scaling gives -4096 for neg nybble, 0 for pos) */ - static unsigned char const right_shifts [16] = { - 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29, - }; - static unsigned char const left_shifts [16] = { - 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11 - }; - int const scale = block_header >> 4; - int const right_shift = right_shifts [scale]; - int const left_shift = left_shifts [scale]; - - /* output position */ - out += BRR_BLOCK_SIZE; - int offset = -BRR_BLOCK_SIZE << 2; - - do /* decode and filter 16 samples */ - { - /* Get nybble, sign-extend, then scale - get byte, select which nybble, sign-extend, then shift based - on scaling. also handles invalid scaling values. */ - int delta = (int) (int8_t) (addr [offset >> 3] << (offset & 4)) - >> right_shift << left_shift; - - out [offset >> 2] = smp2; - - if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */ - { - delta -= smp2 >> 1; - delta += smp2 >> 5; - smp2 = smp1; - delta += smp1; - delta += (-smp1 - (smp1 >> 1)) >> 5; - } - else - { - if ( filter == -4 ) /* mode 0x04 */ - { - delta += smp1 >> 1; - delta += (-smp1) >> 5; - } - else if ( filter > -4 ) /* mode 0x0C */ - { - delta -= smp2 >> 1; - delta += (smp2 + (smp2 >> 1)) >> 4; - delta += smp1; - delta += (-smp1 * 13) >> 7; - } - smp2 = smp1; - } - - delta = CLAMP16( delta ); - smp1 = (int16_t) (delta * 2); /* sign-extend */ - } - while ( (offset += 4) != 0 ); - - /* if next block has end flag set, this block ends early */ - /* (verified) */ - if ( (block_header & 3) != 3 && (*addr & 3) == 1 ) - { - /* skip last 9 samples */ - out -= 9; - goto early_end; - } - } - while ( !(block_header & 1) && addr < RAM + 0x10000 ); - - out [0] = smp2; - out [1] = smp1; - - early_end: - wave_entry->end = (out - 1 - wave_entry->samples) << 12; - - wave_entry->loop = 0; - if ( (block_header & 2) ) - { - if ( loop_start ) - { - int loop = out - loop_start; - wave_entry->loop = loop; - wave_entry->end += 0x3000; - out [2] = loop_start [2]; - out [3] = loop_start [3]; - out [4] = loop_start [4]; - } - else - { - DEBUGF( "loop point outside initial wave\n" ); - } - } - - DEBUGF( "end at %08lx (wave #%d)\n", - (unsigned long)(addr - RAM), raw_voice->waveform ); - - /* add to cache */ - this->wave_entry_old [this->oldsize++] = *wave_entry; -wave_in_cache:; - } -} -#endif - -static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice, - struct src_dir const* const sd, - struct raw_voice_t const* const raw_voice, - const int key_on_delay, const int vbit) ICODE_ATTR; -static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice, - struct src_dir const* const sd, - struct raw_voice_t const* const raw_voice, - const int key_on_delay, const int vbit) { - #undef RAM - #define RAM ram.ram - int const env_rate_init = 0x7800; - voice->key_on_delay = key_on_delay; - if ( key_on_delay == 0 ) - { - this->keys_down |= vbit; - voice->envx = 0; - voice->env_mode = state_attack; - voice->env_timer = env_rate_init; /* TODO: inaccurate? */ - unsigned start_addr = GET_LE16A(sd [raw_voice->waveform].start); - #if !SPC_BRRCACHE - { - voice->addr = RAM + start_addr; - /* BRR filter uses previous samples */ - voice->samples [BRR_BLOCK_SIZE + 1] = 0; - voice->samples [BRR_BLOCK_SIZE + 2] = 0; - /* decode three samples immediately */ - voice->position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1; - voice->block_header = 0; /* "previous" BRR header */ - } - #else - { - voice->position = 3 * 0x1000 - 1; - struct cache_entry_t* const wave_entry = - &this->wave_entry [raw_voice->waveform]; - - /* predecode BRR if not already */ - if ( wave_entry->start_addr != start_addr ) - { - /* the following line can be replaced by the indicated block - in decode_brr() */ - decode_brr( this, start_addr, voice, raw_voice ); - } - - voice->samples = wave_entry->samples; - voice->wave_end = wave_entry->end; - voice->wave_loop = wave_entry->loop; - } - #endif - } -} - -void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf ) -{ - #undef RAM -#ifdef CPU_ARM - uint8_t* const ram_ = ram.ram; - #define RAM ram_ -#else - #define RAM ram.ram -#endif -#if 0 - EXIT_TIMER(cpu); - ENTER_TIMER(dsp); -#endif - - /* Here we check for keys on/off. Docs say that successive writes - to KON/KOF must be separated by at least 2 Ts periods or risk - being neglected. Therefore DSP only looks at these during an - update, and not at the time of the write. Only need to do this - once however, since the regs haven't changed over the whole - period we need to catch up with. */ - - { - int key_ons = this->r.g.key_ons; - int key_offs = this->r.g.key_offs; - /* keying on a voice resets that bit in ENDX */ - this->r.g.wave_ended &= ~key_ons; - /* key_off bits prevent key_on from being acknowledged */ - this->r.g.key_ons = key_ons & key_offs; - - /* process key events outside loop, since they won't re-occur */ - struct voice_t* voice = this->voice_state + 8; - int vbit = 0x80; - do - { - --voice; - if ( key_offs & vbit ) - { - voice->env_mode = state_release; - voice->key_on_delay = 0; - } - else if ( key_ons & vbit ) - { - voice->key_on_delay = 8; - } - } - while ( (vbit >>= 1) != 0 ); - } - - struct src_dir const* const sd = - (struct src_dir*) &RAM [this->r.g.wave_page * 0x100]; - - #ifdef ROCKBOX_BIG_ENDIAN - /* Convert endiannesses before entering loops - these - get used alot */ - const uint32_t rates[VOICE_COUNT] = - { - GET_LE16A( this->r.voice[0].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[1].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[2].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[3].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[4].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[5].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[6].rate ) & 0x3FFF, - GET_LE16A( this->r.voice[7].rate ) & 0x3FFF, - }; - #define VOICE_RATE(x) *(x) - #define IF_RBE(...) __VA_ARGS__ - #ifdef CPU_COLDFIRE - /* Initialize mask register with the buffer address mask */ - asm volatile ("move.l %[m], %%mask" : : [m]"i"(FIR_BUF_MASK)); - const int echo_wrap = (this->r.g.echo_delay & 15) * 0x800; - const int echo_start = this->r.g.echo_page * 0x100; - #endif /* CPU_COLDFIRE */ - #else - #define VOICE_RATE(x) (INT16A(raw_voice->rate) & 0x3FFF) - #define IF_RBE(...) - #endif /* ROCKBOX_BIG_ENDIAN */ - -#if !SPC_NOINTERP - int const slow_gaussian = (this->r.g.pitch_mods >> 1) | - this->r.g.noise_enables; -#endif - /* (g.flags & 0x40) ? 30 : 14 */ - int const global_muting = ((this->r.g.flags & 0x40) >> 2) + 14 - 8; - int const global_vol_0 = this->r.g.volume_0; - int const global_vol_1 = this->r.g.volume_1; - - /* each rate divides exactly into 0x7800 without remainder */ - int const env_rate_init = 0x7800; - static unsigned short const env_rates [0x20] ICONST_ATTR = - { - 0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C, - 0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180, - 0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00, - 0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800 - }; - - do /* one pair of output samples per iteration */ - { - /* Noise */ - if ( this->r.g.noise_enables ) - { - if ( (this->noise_count -= - env_rates [this->r.g.flags & 0x1F]) <= 0 ) - { - this->noise_count = env_rate_init; - int feedback = (this->noise << 13) ^ (this->noise << 14); - this->noise = (feedback & 0x8000) ^ (this->noise >> 1 & ~1); - } - } - -#if !SPC_NOECHO - int echo_0 = 0; - int echo_1 = 0; -#endif - long prev_outx = 0; /* TODO: correct value for first channel? */ - int chans_0 = 0; - int chans_1 = 0; - /* TODO: put raw_voice pointer in voice_t? */ - struct raw_voice_t * raw_voice = this->r.voice; - struct voice_t* voice = this->voice_state; - int vbit = 1; - IF_RBE( const uint32_t* vr = rates; ) - for ( ; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice IF_RBE( , ++vr ) ) - { - /* pregen involves checking keyon, etc */ -#if 0 - ENTER_TIMER(dsp_pregen); -#endif - - /* Key on events are delayed */ - int key_on_delay = voice->key_on_delay; - - if ( --key_on_delay >= 0 ) /* <1% of the time */ - { - key_on(this,voice,sd,raw_voice,key_on_delay,vbit); - } - - if ( !(this->keys_down & vbit) ) /* Silent channel */ - { - silent_chan: - raw_voice->envx = 0; - raw_voice->outx = 0; - prev_outx = 0; - continue; - } - - /* Envelope */ - { - int const ENV_RANGE = 0x800; - int env_mode = voice->env_mode; - int adsr0 = raw_voice->adsr [0]; - int env_timer; - if ( env_mode != state_release ) /* 99% of the time */ - { - env_timer = voice->env_timer; - if ( adsr0 & 0x80 ) /* 79% of the time */ - { - int adsr1 = raw_voice->adsr [1]; - if ( env_mode == state_sustain ) /* 74% of the time */ - { - if ( (env_timer -= env_rates [adsr1 & 0x1F]) > 0 ) - goto write_env_timer; - - int envx = voice->envx; - envx--; /* envx *= 255 / 256 */ - envx -= envx >> 8; - voice->envx = envx; - /* TODO: should this be 8? */ - raw_voice->envx = envx >> 4; - goto init_env_timer; - } - else if ( env_mode < 0 ) /* 25% state_decay */ - { - int envx = voice->envx; - if ( (env_timer -= - env_rates [(adsr0 >> 3 & 0x0E) + 0x10]) <= 0 ) - { - envx--; /* envx *= 255 / 256 */ - envx -= envx >> 8; - voice->envx = envx; - /* TODO: should this be 8? */ - raw_voice->envx = envx >> 4; - env_timer = env_rate_init; - } - - int sustain_level = adsr1 >> 5; - if ( envx <= (sustain_level + 1) * 0x100 ) - voice->env_mode = state_sustain; - - goto write_env_timer; - } - else /* state_attack */ - { - int t = adsr0 & 0x0F; - if ( (env_timer -= env_rates [t * 2 + 1]) > 0 ) - goto write_env_timer; - - int envx = voice->envx; - - int const step = ENV_RANGE / 64; - envx += step; - if ( t == 15 ) - envx += ENV_RANGE / 2 - step; - - if ( envx >= ENV_RANGE ) - { - envx = ENV_RANGE - 1; - voice->env_mode = state_decay; - } - voice->envx = envx; - /* TODO: should this be 8? */ - raw_voice->envx = envx >> 4; - goto init_env_timer; - } - } - else /* gain mode */ - { - int t = raw_voice->gain; - if ( t < 0x80 ) - { - raw_voice->envx = t; - voice->envx = t << 4; - goto env_end; - } - else - { - if ( (env_timer -= env_rates [t & 0x1F]) > 0 ) - goto write_env_timer; - - int envx = voice->envx; - int mode = t >> 5; - if ( mode <= 5 ) /* decay */ - { - int step = ENV_RANGE / 64; - if ( mode == 5 ) /* exponential */ - { - envx--; /* envx *= 255 / 256 */ - step = envx >> 8; - } - if ( (envx -= step) < 0 ) - { - envx = 0; - if ( voice->env_mode == state_attack ) - voice->env_mode = state_decay; - } - } - else /* attack */ - { - int const step = ENV_RANGE / 64; - envx += step; - if ( mode == 7 && - envx >= ENV_RANGE * 3 / 4 + step ) - envx += ENV_RANGE / 256 - step; - - if ( envx >= ENV_RANGE ) - envx = ENV_RANGE - 1; - } - voice->envx = envx; - /* TODO: should this be 8? */ - raw_voice->envx = envx >> 4; - goto init_env_timer; - } - } - } - else /* state_release */ - { - int envx = voice->envx; - if ( (envx -= ENV_RANGE / 256) > 0 ) - { - voice->envx = envx; - raw_voice->envx = envx >> 8; - goto env_end; - } - else - { - /* bit was set, so this clears it */ - this->keys_down ^= vbit; - voice->envx = 0; - goto silent_chan; - } - } - init_env_timer: - env_timer = env_rate_init; - write_env_timer: - voice->env_timer = env_timer; - env_end:; - } -#if 0 - EXIT_TIMER(dsp_pregen); - - ENTER_TIMER(dsp_gen); -#endif - #if !SPC_BRRCACHE - /* Decode BRR block */ - if ( voice->position >= BRR_BLOCK_SIZE * 0x1000 ) - { - voice->position -= BRR_BLOCK_SIZE * 0x1000; - - uint8_t const* addr = voice->addr; - if ( addr >= RAM + 0x10000 ) - addr -= 0x10000; - - /* action based on previous block's header */ - if ( voice->block_header & 1 ) - { - addr = RAM + GET_LE16A( sd [raw_voice->waveform].loop ); - this->r.g.wave_ended |= vbit; - if ( !(voice->block_header & 2) ) /* 1% of the time */ - { - /* first block was end block; - don't play anything (verified) */ - /* bit was set, so this clears it */ - this->keys_down ^= vbit; - - /* since voice->envx is 0, - samples and position don't matter */ - raw_voice->envx = 0; - voice->envx = 0; - goto skip_decode; - } - } - - /* header */ - int const block_header = *addr; - addr += 9; - voice->addr = addr; - voice->block_header = block_header; - int const filter = (block_header & 0x0C) - 0x08; - - /* scaling (invalid scaling gives -4096 for neg nybble, - 0 for pos) */ - static unsigned char const right_shifts [16] = { - 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29, - }; - static unsigned char const left_shifts [16] = { - 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11 - }; - int const scale = block_header >> 4; - int const right_shift = right_shifts [scale]; - int const left_shift = left_shifts [scale]; - - /* previous samples */ - int smp2 = voice->samples [BRR_BLOCK_SIZE + 1]; - int smp1 = voice->samples [BRR_BLOCK_SIZE + 2]; - voice->samples [0] = voice->samples [BRR_BLOCK_SIZE]; - - /* output position */ - short* out = voice->samples + (1 + BRR_BLOCK_SIZE); - int offset = -BRR_BLOCK_SIZE << 2; - - /* if next block has end flag set, - this block ends early (verified) */ - if ( (block_header & 3) != 3 && (*addr & 3) == 1 ) - { - /* arrange for last 9 samples to be skipped */ - int const skip = 9; - out += (skip & 1); - voice->samples [skip] = voice->samples [BRR_BLOCK_SIZE]; - voice->position += skip * 0x1000; - offset = (-BRR_BLOCK_SIZE + (skip & ~1)) << 2; - addr -= skip / 2; - /* force sample to end on next decode */ - voice->block_header = 1; - } - - do /* decode and filter 16 samples */ - { - /* Get nybble, sign-extend, then scale - get byte, select which nybble, sign-extend, then shift - based on scaling. also handles invalid scaling values.*/ - int delta = (int) (int8_t) (addr [offset >> 3] << - (offset & 4)) >> right_shift << left_shift; - - out [offset >> 2] = smp2; - - if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */ - { - delta -= smp2 >> 1; - delta += smp2 >> 5; - smp2 = smp1; - delta += smp1; - delta += (-smp1 - (smp1 >> 1)) >> 5; - } - else - { - if ( filter == -4 ) /* mode 0x04 */ - { - delta += smp1 >> 1; - delta += (-smp1) >> 5; - } - else if ( filter > -4 ) /* mode 0x0C */ - { - delta -= smp2 >> 1; - delta += (smp2 + (smp2 >> 1)) >> 4; - delta += smp1; - delta += (-smp1 * 13) >> 7; - } - smp2 = smp1; - } - - delta = CLAMP16( delta ); - smp1 = (int16_t) (delta * 2); /* sign-extend */ - } - while ( (offset += 4) != 0 ); - - out [0] = smp2; - out [1] = smp1; - - skip_decode:; - } - #endif - - /* Get rate (with possible modulation) */ - int rate = VOICE_RATE(vr); - if ( this->r.g.pitch_mods & vbit ) - rate = (rate * (prev_outx + 32768)) >> 15; - - #if !SPC_NOINTERP - /* Interleved gauss table (to improve cache coherency). */ - /* gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] */ - static short const gauss [512] = - { -370,1305, 366,1305, 362,1304, 358,1304, 354,1304, 351,1304, 347,1304, 343,1303, -339,1303, 336,1303, 332,1302, 328,1302, 325,1301, 321,1300, 318,1300, 314,1299, -311,1298, 307,1297, 304,1297, 300,1296, 297,1295, 293,1294, 290,1293, 286,1292, -283,1291, 280,1290, 276,1288, 273,1287, 270,1286, 267,1284, 263,1283, 260,1282, -257,1280, 254,1279, 251,1277, 248,1275, 245,1274, 242,1272, 239,1270, 236,1269, -233,1267, 230,1265, 227,1263, 224,1261, 221,1259, 218,1257, 215,1255, 212,1253, -210,1251, 207,1248, 204,1246, 201,1244, 199,1241, 196,1239, 193,1237, 191,1234, -188,1232, 186,1229, 183,1227, 180,1224, 178,1221, 175,1219, 173,1216, 171,1213, -168,1210, 166,1207, 163,1205, 161,1202, 159,1199, 156,1196, 154,1193, 152,1190, -150,1186, 147,1183, 145,1180, 143,1177, 141,1174, 139,1170, 137,1167, 134,1164, -132,1160, 130,1157, 128,1153, 126,1150, 124,1146, 122,1143, 120,1139, 118,1136, -117,1132, 115,1128, 113,1125, 111,1121, 109,1117, 107,1113, 106,1109, 104,1106, -102,1102, 100,1098, 99,1094, 97,1090, 95,1086, 94,1082, 92,1078, 90,1074, - 89,1070, 87,1066, 86,1061, 84,1057, 83,1053, 81,1049, 80,1045, 78,1040, - 77,1036, 76,1032, 74,1027, 73,1023, 71,1019, 70,1014, 69,1010, 67,1005, - 66,1001, 65, 997, 64, 992, 62, 988, 61, 983, 60, 978, 59, 974, 58, 969, - 56, 965, 55, 960, 54, 955, 53, 951, 52, 946, 51, 941, 50, 937, 49, 932, - 48, 927, 47, 923, 46, 918, 45, 913, 44, 908, 43, 904, 42, 899, 41, 894, - 40, 889, 39, 884, 38, 880, 37, 875, 36, 870, 36, 865, 35, 860, 34, 855, - 33, 851, 32, 846, 32, 841, 31, 836, 30, 831, 29, 826, 29, 821, 28, 816, - 27, 811, 27, 806, 26, 802, 25, 797, 24, 792, 24, 787, 23, 782, 23, 777, - 22, 772, 21, 767, 21, 762, 20, 757, 20, 752, 19, 747, 19, 742, 18, 737, - 17, 732, 17, 728, 16, 723, 16, 718, 15, 713, 15, 708, 15, 703, 14, 698, - 14, 693, 13, 688, 13, 683, 12, 678, 12, 674, 11, 669, 11, 664, 11, 659, - 10, 654, 10, 649, 10, 644, 9, 640, 9, 635, 9, 630, 8, 625, 8, 620, - 8, 615, 7, 611, 7, 606, 7, 601, 6, 596, 6, 592, 6, 587, 6, 582, - 5, 577, 5, 573, 5, 568, 5, 563, 4, 559, 4, 554, 4, 550, 4, 545, - 4, 540, 3, 536, 3, 531, 3, 527, 3, 522, 3, 517, 2, 513, 2, 508, - 2, 504, 2, 499, 2, 495, 2, 491, 2, 486, 1, 482, 1, 477, 1, 473, - 1, 469, 1, 464, 1, 460, 1, 456, 1, 451, 1, 447, 1, 443, 1, 439, - 0, 434, 0, 430, 0, 426, 0, 422, 0, 418, 0, 414, 0, 410, 0, 405, - 0, 401, 0, 397, 0, 393, 0, 389, 0, 385, 0, 381, 0, 378, 0, 374, - }; - /* Gaussian interpolation using most recent 4 samples */ - long position = voice->position; - voice->position += rate; - short const* interp = voice->samples + (position >> 12); - int offset = position >> 4 & 0xFF; - - /* Only left half of gaussian kernel is in table, so we must mirror - for right half */ - short const* fwd = gauss + offset * 2; - short const* rev = gauss + 510 - offset * 2; - - /* Use faster gaussian interpolation when exact result isn't needed - by pitch modulator of next channel */ - int amp_0, amp_1; - if ( !(slow_gaussian & vbit) ) /* 99% of the time */ - { - /* Main optimization is lack of clamping. Not a problem since - output never goes more than +/- 16 outside 16-bit range and - things are clamped later anyway. Other optimization is to - preserve fractional accuracy, eliminating several masks. */ - int output = (((fwd [0] * interp [0] + - fwd [1] * interp [1] + - rev [1] * interp [2] + - rev [0] * interp [3] ) >> 11) * voice->envx) >> 11; - - /* duplicated here to give compiler more to run in parallel */ - amp_0 = voice->volume [0] * output; - amp_1 = voice->volume [1] * output; - raw_voice->outx = output >> 8; - } - else - { - int output = *(int16_t*) &this->noise; - if ( !(this->r.g.noise_enables & vbit) ) - { - output = (fwd [0] * interp [0]) & ~0xFFF; - output = (output + fwd [1] * interp [1]) & ~0xFFF; - output = (output + rev [1] * interp [2]) >> 12; - output = (int16_t) (output * 2); - output += ((rev [0] * interp [3]) >> 12) * 2; - output = CLAMP16( output ); - } - output = (output * voice->envx) >> 11 & ~1; - - /* duplicated here to give compiler more to run in parallel */ - amp_0 = voice->volume [0] * output; - amp_1 = voice->volume [1] * output; - prev_outx = output; - raw_voice->outx = (int8_t) (output >> 8); - } - #else /* SPCNOINTERP */ - /* two-point linear interpolation */ - #ifdef CPU_COLDFIRE - int amp_0 = (int16_t)this->noise; - int amp_1; - - if ( (this->r.g.noise_enables & vbit) == 0 ) - { - uint32_t f = voice->position; - int32_t y0; - - /** - * Formula (fastest found so far of MANY): - * output = y0 + f*y1 - f*y0 - */ - asm volatile ( - /* separate fractional and whole parts */ - "move.l %[f], %[y1] \r\n" - "and.l #0xfff, %[f] \r\n" - "lsr.l %[sh], %[y1] \r\n" - /* load samples y0 (upper) & y1 (lower) */ - "move.l 2(%[s], %[y1].l*2), %[y1] \r\n" - /* %acc0 = f*y1 */ - "mac.w %[f]l, %[y1]l, %%acc0 \r\n" - /* %acc0 -= f*y0 */ - "msac.w %[f]l, %[y1]u, %%acc0 \r\n" - /* separate out y0 and sign extend */ - "swap %[y1] \r\n" - "movea.w %[y1], %[y0] \r\n" - /* fetch result, scale down and add y0 */ - "movclr.l %%acc0, %[y1] \r\n" - /* output = y0 + (result >> 12) */ - "asr.l %[sh], %[y1] \r\n" - "add.l %[y0], %[y1] \r\n" - : [f]"+d"(f), [y0]"=&a"(y0), [y1]"=&d"(amp_0) - : [s]"a"(voice->samples), [sh]"d"(12) - ); - } - - /* apply voice envelope to output */ - asm volatile ( - "mac.w %[output]l, %[envx]l, %%acc0 \r\n" - : - : [output]"r"(amp_0), [envx]"r"(voice->envx) - ); - - /* advance voice position */ - voice->position += rate; - - /* fetch output, scale and apply left and right - voice volume */ - asm volatile ( - "movclr.l %%acc0, %[output] \r\n" - "asr.l %[sh], %[output] \r\n" - "mac.l %[vvol_0], %[output], %%acc0 \r\n" - "mac.l %[vvol_1], %[output], %%acc1 \r\n" - : [output]"=&d"(amp_0) - : [vvol_0]"r"((int)voice->volume[0]), - [vvol_1]"r"((int)voice->volume[1]), - [sh]"d"(11) - ); - - /* save this output into previous, scale and save in - output register */ - prev_outx = amp_0; - raw_voice->outx = amp_0 >> 8; - - /* fetch final voice output */ - asm volatile ( - "movclr.l %%acc0, %[amp_0] \r\n" - "movclr.l %%acc1, %[amp_1] \r\n" - : [amp_0]"=r"(amp_0), [amp_1]"=r"(amp_1) - ); - #elif defined (CPU_ARM) - int amp_0, amp_1; - - if ( (this->r.g.noise_enables & vbit) != 0 ) { - amp_0 = *(int16_t *)&this->noise; - } else { - uint32_t f = voice->position; - amp_0 = (uint32_t)voice->samples; - - asm volatile( - "mov %[y1], %[f], lsr #12 \r\n" - "eor %[f], %[f], %[y1], lsl #12 \r\n" - "add %[y1], %[y0], %[y1], lsl #1 \r\n" - "ldrsh %[y0], [%[y1], #2] \r\n" - "ldrsh %[y1], [%[y1], #4] \r\n" - "sub %[y1], %[y1], %[y0] \r\n" - "mul %[f], %[y1], %[f] \r\n" - "add %[y0], %[y0], %[f], asr #12 \r\n" - : [f]"+r"(f), [y0]"+r"(amp_0), [y1]"=&r"(amp_1) - ); - } - - voice->position += rate; - - asm volatile( - "mul %[amp_1], %[amp_0], %[envx] \r\n" - "mov %[amp_0], %[amp_1], asr #11 \r\n" - "mov %[amp_1], %[amp_0], asr #8 \r\n" - : [amp_0]"+r"(amp_0), [amp_1]"=&r"(amp_1) - : [envx]"r"(voice->envx) - ); - - prev_outx = amp_0; - raw_voice->outx = (int8_t)amp_1; - - asm volatile( - "mul %[amp_1], %[amp_0], %[vol_1] \r\n" - "mul %[amp_0], %[vol_0], %[amp_0] \r\n" - : [amp_0]"+r"(amp_0), [amp_1]"+r"(amp_1) - : [vol_0]"r"((int)voice->volume[0]), - [vol_1]"r"((int)voice->volume[1]) - ); - #else /* Unoptimized CPU */ - int output; - - if ( (this->r.g.noise_enables & vbit) == 0 ) - { - int const fraction = voice->position & 0xfff; - short const* const pos = (voice->samples + (voice->position >> 12)) + 1; - output = pos[0] + ((fraction * (pos[1] - pos[0])) >> 12); - } else { - output = *(int16_t *)&this->noise; - } - - voice->position += rate; - - output = (output * voice->envx) >> 11; - - /* duplicated here to give compiler more to run in parallel */ - int amp_0 = voice->volume [0] * output; - int amp_1 = voice->volume [1] * output; - - prev_outx = output; - raw_voice->outx = (int8_t) (output >> 8); - #endif /* CPU_* */ - #endif /* SPCNOINTERP */ - - #if SPC_BRRCACHE - if ( voice->position >= voice->wave_end ) - { - long loop_len = voice->wave_loop << 12; - voice->position -= loop_len; - this->r.g.wave_ended |= vbit; - if ( !loop_len ) - { - this->keys_down ^= vbit; - raw_voice->envx = 0; - voice->envx = 0; - } - } - #endif -#if 0 - EXIT_TIMER(dsp_gen); - - ENTER_TIMER(dsp_mix); -#endif - chans_0 += amp_0; - chans_1 += amp_1; - #if !SPC_NOECHO - if ( this->r.g.echo_ons & vbit ) - { - echo_0 += amp_0; - echo_1 += amp_1; - } - #endif -#if 0 - EXIT_TIMER(dsp_mix); -#endif - } - /* end of voice loop */ - - #if !SPC_NOECHO - #ifdef CPU_COLDFIRE - /* Read feedback from echo buffer */ - int echo_pos = this->echo_pos; - uint8_t* const echo_ptr = RAM + ((echo_start + echo_pos) & 0xFFFF); - echo_pos += 4; - if ( echo_pos >= echo_wrap ) - echo_pos = 0; - this->echo_pos = echo_pos; - int fb = swap_odd_even32(*(int32_t *)echo_ptr); - int out_0, out_1; - - /* Keep last 8 samples */ - *this->last_fir_ptr = fb; - this->last_fir_ptr = this->fir_ptr; - - /* Apply echo FIR filter to output samples read from echo buffer - - circular buffer is hardware incremented and masked; FIR - coefficients and buffer history are loaded in parallel with - multiply accumulate operations. Shift left by one here and once - again when calculating feedback to have sample values justified - to bit 31 in the output to ease endian swap, interleaving and - clamping before placing result in the program's echo buffer. */ - int _0, _1, _2; - asm volatile ( - "move.l (%[fir_c]) , %[_2] \r\n" - "mac.w %[fb]u, %[_2]u, <<, (%[fir_p])+&, %[_0], %%acc0 \r\n" - "mac.w %[fb]l, %[_2]u, <<, (%[fir_p])& , %[_1], %%acc1 \r\n" - "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n" - "mac.w %[_0]l, %[_2]l, <<, 4(%[fir_c]) , %[_2], %%acc1 \r\n" - "mac.w %[_1]u, %[_2]u, <<, 4(%[fir_p])& , %[_0], %%acc0 \r\n" - "mac.w %[_1]l, %[_2]u, <<, 8(%[fir_p])& , %[_1], %%acc1 \r\n" - "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n" - "mac.w %[_0]l, %[_2]l, <<, 8(%[fir_c]) , %[_2], %%acc1 \r\n" - "mac.w %[_1]u, %[_2]u, <<, 12(%[fir_p])& , %[_0], %%acc0 \r\n" - "mac.w %[_1]l, %[_2]u, <<, 16(%[fir_p])& , %[_1], %%acc1 \r\n" - "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n" - "mac.w %[_0]l, %[_2]l, <<, 12(%[fir_c]) , %[_2], %%acc1 \r\n" - "mac.w %[_1]u, %[_2]u, <<, 20(%[fir_p])& , %[_0], %%acc0 \r\n" - "mac.w %[_1]l, %[_2]u, << , %%acc1 \r\n" - "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n" - "mac.w %[_0]l, %[_2]l, << , %%acc1 \r\n" - : [_0]"=&r"(_0), [_1]"=&r"(_1), [_2]"=&r"(_2), - [fir_p]"+a"(this->fir_ptr) - : [fir_c]"a"(this->fir_coeff), [fb]"r"(fb) - ); - - /* Generate output */ - asm volatile ( - /* fetch filter results _after_ gcc loads asm - block parameters to eliminate emac stalls */ - "movclr.l %%acc0, %[out_0] \r\n" - "movclr.l %%acc1, %[out_1] \r\n" - /* apply global volume */ - "mac.l %[chans_0], %[gv_0] , %%acc2 \r\n" - "mac.l %[chans_1], %[gv_1] , %%acc3 \r\n" - /* apply echo volume and add to final output */ - "mac.l %[ev_0], %[out_0], >>, %%acc2 \r\n" - "mac.l %[ev_1], %[out_1], >>, %%acc3 \r\n" - : [out_0]"=&r"(out_0), [out_1]"=&r"(out_1) - : [chans_0]"r"(chans_0), [gv_0]"r"(global_vol_0), - [ev_0]"r"((int)this->r.g.echo_volume_0), - [chans_1]"r"(chans_1), [gv_1]"r"(global_vol_1), - [ev_1]"r"((int)this->r.g.echo_volume_1) - ); - - /* Feedback into echo buffer */ - if ( !(this->r.g.flags & 0x20) ) - { - asm volatile ( - /* scale echo voices; saturate if overflow */ - "mac.l %[sh], %[e1] , %%acc1 \r\n" - "mac.l %[sh], %[e0] , %%acc0 \r\n" - /* add scaled output from FIR filter */ - "mac.l %[out_1], %[ef], <<, %%acc1 \r\n" - "mac.l %[out_0], %[ef], <<, %%acc0 \r\n" - /* swap and fetch feedback results - simply - swap_odd_even32 mixed in between macs and - movclrs to mitigate stall issues */ - "move.l #0x00ff00ff, %[sh] \r\n" - "movclr.l %%acc1, %[e1] \r\n" - "swap %[e1] \r\n" - "movclr.l %%acc0, %[e0] \r\n" - "move.w %[e1], %[e0] \r\n" - "and.l %[e0], %[sh] \r\n" - "eor.l %[sh], %[e0] \r\n" - "lsl.l #8, %[sh] \r\n" - "lsr.l #8, %[e0] \r\n" - "or.l %[sh], %[e0] \r\n" - /* save final feedback into echo buffer */ - "move.l %[e0], (%[echo_ptr]) \r\n" - : [e0]"+d"(echo_0), [e1]"+d"(echo_1) - : [out_0]"r"(out_0), [out_1]"r"(out_1), - [ef]"r"((int)this->r.g.echo_feedback), - [echo_ptr]"a"((int32_t *)echo_ptr), - [sh]"d"(1 << 9) - ); - } - - /* Output final samples */ - asm volatile ( - /* fetch output saved in %acc2 and %acc3 */ - "movclr.l %%acc2, %[out_0] \r\n" - "movclr.l %%acc3, %[out_1] \r\n" - /* scale right by global_muting shift */ - "asr.l %[gm], %[out_0] \r\n" - "asr.l %[gm], %[out_1] \r\n" - : [out_0]"=&d"(out_0), [out_1]"=&d"(out_1) - : [gm]"d"(global_muting) - ); - - out_buf [ 0] = out_0; - out_buf [WAV_CHUNK_SIZE] = out_1; - out_buf ++; - #elif defined (CPU_ARM) - /* Read feedback from echo buffer */ - int echo_pos = this->echo_pos; - uint8_t* const echo_ptr = RAM + - ((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF); - echo_pos += 4; - if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 ) - echo_pos = 0; - this->echo_pos = echo_pos; - - int fb_0 = GET_LE16SA( echo_ptr ); - int fb_1 = GET_LE16SA( echo_ptr + 2 ); - - /* Keep last 8 samples */ - int32_t *fir_ptr = this->fir_ptr; - - /* Apply FIR */ - asm volatile ( - "str %[fb_0], [%[fir_p]], #4 \r\n" - "str %[fb_1], [%[fir_p]], #4 \r\n" - /* duplicate at +8 eliminates wrap checking below */ - "str %[fb_0], [%[fir_p], #56] \r\n" - "str %[fb_1], [%[fir_p], #60] \r\n" - : [fir_p]"+r"(fir_ptr) - : [fb_0]"r"(fb_0), [fb_1]"r"(fb_1) - ); - - this->fir_ptr = (int32_t *)((intptr_t)fir_ptr & FIR_BUF_MASK); - int32_t *fir_coeff = this->fir_coeff; - - asm volatile ( - "ldmia %[fir_c]!, { r0-r1 } \r\n" - "ldmia %[fir_p]!, { r4-r5 } \r\n" - "mul %[fb_0], r0, %[fb_0] \r\n" - "mul %[fb_1], r0, %[fb_1] \r\n" - "mla %[fb_0], r4, r1, %[fb_0] \r\n" - "mla %[fb_1], r5, r1, %[fb_1] \r\n" - "ldmia %[fir_c]!, { r0-r1 } \r\n" - "ldmia %[fir_p]!, { r2-r5 } \r\n" - "mla %[fb_0], r2, r0, %[fb_0] \r\n" - "mla %[fb_1], r3, r0, %[fb_1] \r\n" - "mla %[fb_0], r4, r1, %[fb_0] \r\n" - "mla %[fb_1], r5, r1, %[fb_1] \r\n" - "ldmia %[fir_c]!, { r0-r1 } \r\n" - "ldmia %[fir_p]!, { r2-r5 } \r\n" - "mla %[fb_0], r2, r0, %[fb_0] \r\n" - "mla %[fb_1], r3, r0, %[fb_1] \r\n" - "mla %[fb_0], r4, r1, %[fb_0] \r\n" - "mla %[fb_1], r5, r1, %[fb_1] \r\n" - "ldmia %[fir_c]!, { r0-r1 } \r\n" - "ldmia %[fir_p]!, { r2-r5 } \r\n" - "mla %[fb_0], r2, r0, %[fb_0] \r\n" - "mla %[fb_1], r3, r0, %[fb_1] \r\n" - "mla %[fb_0], r4, r1, %[fb_0] \r\n" - "mla %[fb_1], r5, r1, %[fb_1] \r\n" - : [fb_0]"+r"(fb_0), [fb_1]"+r"(fb_1), - [fir_p]"+r"(fir_ptr), [fir_c]"+r"(fir_coeff) - : - : "r0", "r1", "r2", "r3", "r4", "r5" - ); - - /* Generate output */ - int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0) - >> global_muting; - int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1) - >> global_muting; - - out_buf [ 0] = amp_0; - out_buf [WAV_CHUNK_SIZE] = amp_1; - out_buf ++; - - if ( !(this->r.g.flags & 0x20) ) - { - /* Feedback into echo buffer */ - int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14); - int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14); - e0 = CLAMP16( e0 ); - SET_LE16A( echo_ptr , e0 ); - e1 = CLAMP16( e1 ); - SET_LE16A( echo_ptr + 2, e1 ); - } - #else /* Unoptimized CPU */ - /* Read feedback from echo buffer */ - int echo_pos = this->echo_pos; - uint8_t* const echo_ptr = RAM + - ((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF); - echo_pos += 4; - if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 ) - echo_pos = 0; - this->echo_pos = echo_pos; - int fb_0 = GET_LE16SA( echo_ptr ); - int fb_1 = GET_LE16SA( echo_ptr + 2 ); - - /* Keep last 8 samples */ - int (* const fir_ptr) [2] = this->fir_buf + this->fir_pos; - this->fir_pos = (this->fir_pos + 1) & (FIR_BUF_HALF - 1); - fir_ptr [ 0] [0] = fb_0; - fir_ptr [ 0] [1] = fb_1; - /* duplicate at +8 eliminates wrap checking below */ - fir_ptr [FIR_BUF_HALF] [0] = fb_0; - fir_ptr [FIR_BUF_HALF] [1] = fb_1; - - /* Apply FIR */ - fb_0 *= this->fir_coeff [0]; - fb_1 *= this->fir_coeff [0]; - - #define DO_PT( i )\ - fb_0 += fir_ptr [i] [0] * this->fir_coeff [i];\ - fb_1 += fir_ptr [i] [1] * this->fir_coeff [i]; - - DO_PT( 1 ) - DO_PT( 2 ) - DO_PT( 3 ) - DO_PT( 4 ) - DO_PT( 5 ) - DO_PT( 6 ) - DO_PT( 7 ) - - /* Generate output */ - int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0) - >> global_muting; - int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1) - >> global_muting; - out_buf [ 0] = amp_0; - out_buf [WAV_CHUNK_SIZE] = amp_1; - out_buf ++; - - if ( !(this->r.g.flags & 0x20) ) - { - /* Feedback into echo buffer */ - int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14); - int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14); - e0 = CLAMP16( e0 ); - SET_LE16A( echo_ptr , e0 ); - e1 = CLAMP16( e1 ); - SET_LE16A( echo_ptr + 2, e1 ); - } - #endif /* CPU_* */ - #else /* SPCNOECHO == 1*/ - /* Generate output */ - int amp_0 = (chans_0 * global_vol_0) >> global_muting; - int amp_1 = (chans_1 * global_vol_1) >> global_muting; - out_buf [ 0] = amp_0; - out_buf [WAV_CHUNK_SIZE] = amp_1; - out_buf ++; - #endif /* SPCNOECHO */ - } - while ( --count ); -#if 0 - EXIT_TIMER(dsp); - ENTER_TIMER(cpu); -#endif -} - -void DSP_reset( struct Spc_Dsp* this ) -{ - this->keys_down = 0; - this->echo_pos = 0; - this->noise_count = 0; - this->noise = 2; - - this->r.g.flags = 0xE0; /* reset, mute, echo off */ - this->r.g.key_ons = 0; - - ci->memset( this->voice_state, 0, sizeof this->voice_state ); - - int i; - for ( i = VOICE_COUNT; --i >= 0; ) - { - struct voice_t* v = this->voice_state + i; - v->env_mode = state_release; - v->addr = ram.ram; - } - - #if SPC_BRRCACHE - this->oldsize = 0; - for ( i = 0; i < 256; i++ ) - this->wave_entry [i].start_addr = -1; - #endif - -#if defined(CPU_COLDFIRE) - this->fir_ptr = fir_buf; - this->last_fir_ptr = &fir_buf [7]; - ci->memset( fir_buf, 0, sizeof fir_buf ); -#elif defined (CPU_ARM) - this->fir_ptr = fir_buf; - ci->memset( fir_buf, 0, sizeof fir_buf ); -#else - this->fir_pos = 0; - ci->memset( this->fir_buf, 0, sizeof this->fir_buf ); -#endif - - assert( offsetof (struct globals_t,unused9 [2]) == REGISTER_COUNT ); - assert( sizeof (this->r.voice) == REGISTER_COUNT ); -} |