| /*************************************************************************** |
| * __________ __ ___. |
| * Open \______ \ ____ ____ | | _\_ |__ _______ ___ |
| * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / |
| * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < |
| * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ |
| * \/ \/ \/ \/ \/ |
| * $Id$ |
| * |
| * Copyright (C) 2006-2007 Adam Gashlin (hcs) |
| * Copyright (C) 2004-2007 Shay Green (blargg) |
| * Copyright (C) 2002 Brad Martin |
| * |
| * All files in this archive are subject to the GNU General Public License. |
| * See the file COPYING in the source tree root for full license agreement. |
| * |
| * 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]; |
| #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 ); |
| } |