13f08d70fd
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23784 a1c6a512-1295-4272-9138-f99709370657
1276 lines
50 KiB
C
1276 lines
50 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2007-2008 Michael Sevakis (jhMikeS)
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* Copyright (C) 2006-2007 Adam Gashlin (hcs)
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* Copyright (C) 2004-2007 Shay Green (blargg)
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* Copyright (C) 2002 Brad Martin
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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/* The DSP portion (awe!) */
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#include "codeclib.h"
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#include "spc_codec.h"
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#include "spc_profiler.h"
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#if defined(CPU_COLDFIRE) || defined (CPU_ARM)
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int32_t fir_buf[FIR_BUF_CNT]
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__attribute__ ((aligned (FIR_BUF_ALIGN*1))) IBSS_ATTR;
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#endif
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#if SPC_BRRCACHE
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/* a little extra for samples that go past end */
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int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR;
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#endif
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void DSP_write( struct Spc_Dsp* this, int i, int data )
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{
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assert( (unsigned) i < REGISTER_COUNT );
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this->r.reg [i] = data;
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int high = i >> 4;
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int low = i & 0x0F;
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if ( low < 2 ) /* voice volumes */
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{
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int left = *(int8_t const*) &this->r.reg [i & ~1];
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int right = *(int8_t const*) &this->r.reg [i | 1];
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struct voice_t* v = this->voice_state + high;
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v->volume [0] = left;
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v->volume [1] = right;
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}
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else if ( low == 0x0F ) /* fir coefficients */
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{
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this->fir_coeff [7 - high] = (int8_t) data; /* sign-extend */
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}
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}
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/* if ( n < -32768 ) out = -32768; */
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/* if ( n > 32767 ) out = 32767; */
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#define CLAMP16( n ) \
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({ \
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if ( (int16_t) n != n ) \
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n = 0x7FFF ^ (n >> 31); \
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n; \
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})
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#if SPC_BRRCACHE
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static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
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struct voice_t* voice,
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struct raw_voice_t const* const raw_voice ) ICODE_ATTR;
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static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
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struct voice_t* voice,
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struct raw_voice_t const* const raw_voice )
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{
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/* setup same variables as where decode_brr() is called from */
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#undef RAM
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#define RAM ram.ram
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struct src_dir const* const sd =
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&ram.sd[this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
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struct cache_entry_t* const wave_entry =
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&this->wave_entry [raw_voice->waveform];
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/* the following block can be put in place of the call to
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decode_brr() below
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*/
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{
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DEBUGF( "decode at %08x (wave #%d)\n",
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start_addr, raw_voice->waveform );
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/* see if in cache */
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int i;
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for ( i = 0; i < this->oldsize; i++ )
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{
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struct cache_entry_t* e = &this->wave_entry_old [i];
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if ( e->start_addr == start_addr )
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{
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DEBUGF( "found in wave_entry_old (oldsize=%d)\n",
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this->oldsize );
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*wave_entry = *e;
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goto wave_in_cache;
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}
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}
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wave_entry->start_addr = start_addr;
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uint8_t const* const loop_ptr =
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RAM + letoh16(sd[raw_voice->waveform].loop);
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short* loop_start = 0;
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short* out = BRRcache + start_addr * 2;
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wave_entry->samples = out;
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*out++ = 0;
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int smp1 = 0;
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int smp2 = 0;
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uint8_t const* addr = RAM + start_addr;
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int block_header;
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do
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{
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if ( addr == loop_ptr )
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{
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loop_start = out;
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DEBUGF( "loop at %08lx (wave #%d)\n",
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(unsigned long)(addr - RAM), raw_voice->waveform );
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}
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/* header */
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block_header = *addr;
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addr += 9;
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voice->addr = addr;
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int const filter = (block_header & 0x0C) - 0x08;
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/* scaling
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(invalid scaling gives -4096 for neg nybble, 0 for pos) */
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static unsigned char const right_shifts [16] = {
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5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29,
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};
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static unsigned char const left_shifts [16] = {
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0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11
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};
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int const scale = block_header >> 4;
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int const right_shift = right_shifts [scale];
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int const left_shift = left_shifts [scale];
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/* output position */
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out += BRR_BLOCK_SIZE;
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int offset = -BRR_BLOCK_SIZE << 2;
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do /* decode and filter 16 samples */
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{
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/* Get nybble, sign-extend, then scale
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get byte, select which nybble, sign-extend, then shift based
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on scaling. also handles invalid scaling values. */
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int delta = (int) (int8_t) (addr [offset >> 3] << (offset & 4))
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>> right_shift << left_shift;
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out [offset >> 2] = smp2;
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if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */
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{
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delta -= smp2 >> 1;
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delta += smp2 >> 5;
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smp2 = smp1;
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delta += smp1;
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delta += (-smp1 - (smp1 >> 1)) >> 5;
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}
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else
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{
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if ( filter == -4 ) /* mode 0x04 */
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{
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delta += smp1 >> 1;
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delta += (-smp1) >> 5;
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}
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else if ( filter > -4 ) /* mode 0x0C */
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{
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delta -= smp2 >> 1;
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delta += (smp2 + (smp2 >> 1)) >> 4;
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delta += smp1;
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delta += (-smp1 * 13) >> 7;
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}
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smp2 = smp1;
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}
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delta = CLAMP16( delta );
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smp1 = (int16_t) (delta * 2); /* sign-extend */
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}
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while ( (offset += 4) != 0 );
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/* if next block has end flag set, this block ends early */
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/* (verified) */
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if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
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{
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/* skip last 9 samples */
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out -= 9;
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goto early_end;
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}
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}
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while ( !(block_header & 1) && addr < RAM + 0x10000 );
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out [0] = smp2;
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out [1] = smp1;
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early_end:
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wave_entry->end = (out - 1 - wave_entry->samples) << 12;
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wave_entry->loop = 0;
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if ( (block_header & 2) )
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{
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if ( loop_start )
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{
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int loop = out - loop_start;
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wave_entry->loop = loop;
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wave_entry->end += 0x3000;
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out [2] = loop_start [2];
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out [3] = loop_start [3];
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out [4] = loop_start [4];
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}
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else
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{
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DEBUGF( "loop point outside initial wave\n" );
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}
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}
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DEBUGF( "end at %08lx (wave #%d)\n",
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(unsigned long)(addr - RAM), raw_voice->waveform );
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/* add to cache */
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this->wave_entry_old [this->oldsize++] = *wave_entry;
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wave_in_cache:;
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}
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}
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#endif
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static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice,
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struct src_dir const* const sd,
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struct raw_voice_t const* const raw_voice,
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const int key_on_delay, const int vbit) ICODE_ATTR;
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static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice,
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struct src_dir const* const sd,
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struct raw_voice_t const* const raw_voice,
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const int key_on_delay, const int vbit) {
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#undef RAM
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#define RAM ram.ram
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int const env_rate_init = 0x7800;
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voice->key_on_delay = key_on_delay;
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if ( key_on_delay == 0 )
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{
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this->keys_down |= vbit;
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voice->envx = 0;
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voice->env_mode = state_attack;
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voice->env_timer = env_rate_init; /* TODO: inaccurate? */
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unsigned start_addr = letoh16(sd[raw_voice->waveform].start);
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#if !SPC_BRRCACHE
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{
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voice->addr = RAM + start_addr;
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/* BRR filter uses previous samples */
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voice->samples [BRR_BLOCK_SIZE + 1] = 0;
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voice->samples [BRR_BLOCK_SIZE + 2] = 0;
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/* decode three samples immediately */
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voice->position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1;
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voice->block_header = 0; /* "previous" BRR header */
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}
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#else
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{
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voice->position = 3 * 0x1000 - 1;
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struct cache_entry_t* const wave_entry =
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&this->wave_entry [raw_voice->waveform];
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/* predecode BRR if not already */
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if ( wave_entry->start_addr != start_addr )
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{
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/* the following line can be replaced by the indicated block
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in decode_brr() */
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decode_brr( this, start_addr, voice, raw_voice );
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}
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voice->samples = wave_entry->samples;
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voice->wave_end = wave_entry->end;
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voice->wave_loop = wave_entry->loop;
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}
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#endif
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}
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}
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void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
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{
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#undef RAM
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#ifdef CPU_ARM
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uint8_t* const ram_ = ram.ram;
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#define RAM ram_
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#else
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#define RAM ram.ram
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#endif
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#if 0
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EXIT_TIMER(cpu);
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ENTER_TIMER(dsp);
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#endif
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/* Here we check for keys on/off. Docs say that successive writes
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to KON/KOF must be separated by at least 2 Ts periods or risk
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being neglected. Therefore DSP only looks at these during an
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update, and not at the time of the write. Only need to do this
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once however, since the regs haven't changed over the whole
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period we need to catch up with. */
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{
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int key_ons = this->r.g.key_ons;
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int key_offs = this->r.g.key_offs;
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/* keying on a voice resets that bit in ENDX */
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this->r.g.wave_ended &= ~key_ons;
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/* key_off bits prevent key_on from being acknowledged */
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this->r.g.key_ons = key_ons & key_offs;
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/* process key events outside loop, since they won't re-occur */
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struct voice_t* voice = this->voice_state + 8;
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int vbit = 0x80;
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do
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{
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--voice;
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if ( key_offs & vbit )
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{
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voice->env_mode = state_release;
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voice->key_on_delay = 0;
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}
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else if ( key_ons & vbit )
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{
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voice->key_on_delay = 8;
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}
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}
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while ( (vbit >>= 1) != 0 );
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}
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struct src_dir const* const sd =
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&ram.sd[this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
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#ifdef ROCKBOX_BIG_ENDIAN
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/* Convert endiannesses before entering loops - these
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get used alot */
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const uint32_t rates[VOICE_COUNT] =
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{
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GET_LE16A( this->r.voice[0].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[1].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[2].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[3].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[4].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[5].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[6].rate ) & 0x3FFF,
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GET_LE16A( this->r.voice[7].rate ) & 0x3FFF,
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};
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#define VOICE_RATE(x) *(x)
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#define IF_RBE(...) __VA_ARGS__
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#ifdef CPU_COLDFIRE
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/* Initialize mask register with the buffer address mask */
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asm volatile ("move.l %[m], %%mask" : : [m]"i"(FIR_BUF_MASK));
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const int echo_wrap = (this->r.g.echo_delay & 15) * 0x800;
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const int echo_start = this->r.g.echo_page * 0x100;
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#endif /* CPU_COLDFIRE */
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#else
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#define VOICE_RATE(x) (GET_LE16(raw_voice->rate) & 0x3FFF)
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#define IF_RBE(...)
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#endif /* ROCKBOX_BIG_ENDIAN */
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#if !SPC_NOINTERP
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int const slow_gaussian = (this->r.g.pitch_mods >> 1) |
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this->r.g.noise_enables;
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#endif
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/* (g.flags & 0x40) ? 30 : 14 */
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int const global_muting = ((this->r.g.flags & 0x40) >> 2) + 14 - 8;
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int const global_vol_0 = this->r.g.volume_0;
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int const global_vol_1 = this->r.g.volume_1;
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/* each rate divides exactly into 0x7800 without remainder */
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int const env_rate_init = 0x7800;
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static unsigned short const env_rates [0x20] ICONST_ATTR =
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{
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0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C,
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0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180,
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0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00,
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0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800
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};
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do /* one pair of output samples per iteration */
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{
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/* Noise */
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if ( this->r.g.noise_enables )
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{
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if ( (this->noise_count -=
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env_rates [this->r.g.flags & 0x1F]) <= 0 )
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{
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this->noise_count = env_rate_init;
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int feedback = (this->noise << 13) ^ (this->noise << 14);
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this->noise = (feedback & 0x8000) ^ (this->noise >> 1 & ~1);
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}
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}
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|
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#if !SPC_NOECHO
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int echo_0 = 0;
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int echo_1 = 0;
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#endif
|
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long prev_outx = 0; /* TODO: correct value for first channel? */
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int chans_0 = 0;
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int chans_1 = 0;
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/* TODO: put raw_voice pointer in voice_t? */
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struct raw_voice_t * raw_voice = this->r.voice;
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struct voice_t* voice = this->voice_state;
|
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int vbit = 1;
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IF_RBE( const uint32_t* vr = rates; )
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for ( ; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice IF_RBE( , ++vr ) )
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{
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/* pregen involves checking keyon, etc */
|
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#if 0
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ENTER_TIMER(dsp_pregen);
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#endif
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|
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/* Key on events are delayed */
|
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int key_on_delay = voice->key_on_delay;
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|
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if ( --key_on_delay >= 0 ) /* <1% of the time */
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{
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key_on(this,voice,sd,raw_voice,key_on_delay,vbit);
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}
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|
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if ( !(this->keys_down & vbit) ) /* Silent channel */
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{
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silent_chan:
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raw_voice->envx = 0;
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raw_voice->outx = 0;
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prev_outx = 0;
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continue;
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}
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|
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/* Envelope */
|
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{
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int const ENV_RANGE = 0x800;
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int env_mode = voice->env_mode;
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int adsr0 = raw_voice->adsr [0];
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int env_timer;
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if ( env_mode != state_release ) /* 99% of the time */
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{
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env_timer = voice->env_timer;
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if ( adsr0 & 0x80 ) /* 79% of the time */
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{
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int adsr1 = raw_voice->adsr [1];
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if ( env_mode == state_sustain ) /* 74% of the time */
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{
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if ( (env_timer -= env_rates [adsr1 & 0x1F]) > 0 )
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goto write_env_timer;
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|
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int envx = voice->envx;
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envx--; /* envx *= 255 / 256 */
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envx -= envx >> 8;
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voice->envx = envx;
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/* TODO: should this be 8? */
|
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raw_voice->envx = envx >> 4;
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goto init_env_timer;
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}
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else if ( env_mode < 0 ) /* 25% state_decay */
|
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{
|
|
int envx = voice->envx;
|
|
if ( (env_timer -=
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env_rates [(adsr0 >> 3 & 0x0E) + 0x10]) <= 0 )
|
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{
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envx--; /* envx *= 255 / 256 */
|
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envx -= envx >> 8;
|
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voice->envx = envx;
|
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/* TODO: should this be 8? */
|
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raw_voice->envx = envx >> 4;
|
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env_timer = env_rate_init;
|
|
}
|
|
|
|
int sustain_level = adsr1 >> 5;
|
|
if ( envx <= (sustain_level + 1) * 0x100 )
|
|
voice->env_mode = state_sustain;
|
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|
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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 + letoh16(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 );
|
|
}
|