Mercurial > audlegacy-plugins
view src/console/Ay_Apu.cxx @ 3156:08a91bda0829
adapt to audlegacy
author | Yoshiki Yazawa <yaz@honeyplanet.jp> |
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date | Fri, 22 May 2009 19:11:38 +0900 |
parents | 986f098da058 |
children |
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// Game_Music_Emu 0.5.2. http://www.slack.net/~ant/ #include "Ay_Apu.h" /* Copyright (C) 2006 Shay Green. This module is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version. This module is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details. You should have received a copy of the GNU Lesser General Public License along with this module; if not, write to the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "blargg_source.h" // Emulation inaccuracies: // * Noise isn't run when not in use // * Changes to envelope and noise periods are delayed until next reload // * Super-sonic tone should attenuate output to about 60%, not 50% // Tones above this frequency are treated as disabled tone at half volume. // Power of two is more efficient (avoids division). unsigned const inaudible_freq = 16384; int const period_factor = 16; static byte const amp_table [16] = { #define ENTRY( n ) byte (n * Ay_Apu::amp_range + 0.5) // With channels tied together and 1K resistor to ground (as datasheet recommends), // output nearly matches logarithmic curve as claimed. Approx. 1.5 dB per step. ENTRY(0.000000),ENTRY(0.007813),ENTRY(0.011049),ENTRY(0.015625), ENTRY(0.022097),ENTRY(0.031250),ENTRY(0.044194),ENTRY(0.062500), ENTRY(0.088388),ENTRY(0.125000),ENTRY(0.176777),ENTRY(0.250000), ENTRY(0.353553),ENTRY(0.500000),ENTRY(0.707107),ENTRY(1.000000), /* // Measured from an AY-3-8910A chip with date code 8611. // Direct voltages without any load (very linear) ENTRY(0.000000),ENTRY(0.046237),ENTRY(0.064516),ENTRY(0.089785), ENTRY(0.124731),ENTRY(0.173118),ENTRY(0.225806),ENTRY(0.329032), ENTRY(0.360215),ENTRY(0.494624),ENTRY(0.594624),ENTRY(0.672043), ENTRY(0.766129),ENTRY(0.841935),ENTRY(0.926882),ENTRY(1.000000), // With only some load ENTRY(0.000000),ENTRY(0.011940),ENTRY(0.017413),ENTRY(0.024876), ENTRY(0.036318),ENTRY(0.054229),ENTRY(0.072637),ENTRY(0.122388), ENTRY(0.174129),ENTRY(0.239303),ENTRY(0.323881),ENTRY(0.410945), ENTRY(0.527363),ENTRY(0.651741),ENTRY(0.832338),ENTRY(1.000000), */ #undef ENTRY }; static byte const modes [8] = { #define MODE( a0,a1, b0,b1, c0,c1 ) \ (a0 | a1<<1 | b0<<2 | b1<<3 | c0<<4 | c1<<5) MODE( 1,0, 1,0, 1,0 ), MODE( 1,0, 0,0, 0,0 ), MODE( 1,0, 0,1, 1,0 ), MODE( 1,0, 1,1, 1,1 ), MODE( 0,1, 0,1, 0,1 ), MODE( 0,1, 1,1, 1,1 ), MODE( 0,1, 1,0, 0,1 ), MODE( 0,1, 0,0, 0,0 ), }; Ay_Apu::Ay_Apu() { // build full table of the upper 8 envelope waveforms for ( int m = 8; m--; ) { byte* out = env.modes [m]; int flags = modes [m]; for ( int x = 3; --x >= 0; ) { int amp = flags & 1; int end = flags >> 1 & 1; int step = end - amp; amp *= 15; for ( int y = 16; --y >= 0; ) { *out++ = amp_table [amp]; amp += step; } flags >>= 2; } } output( 0 ); volume( 1.0 ); reset(); } void Ay_Apu::reset() { last_time = 0; noise.delay = 0; noise.lfsr = 1; osc_t* osc = &oscs [osc_count]; do { osc--; osc->period = period_factor; osc->delay = 0; osc->last_amp = 0; osc->phase = 0; } while ( osc != oscs ); for ( int i = sizeof regs; --i >= 0; ) regs [i] = 0; regs [7] = 0xFF; write_data_( 13, 0 ); } void Ay_Apu::write_data_( int addr, int data ) { assert( (unsigned) addr < reg_count ); if ( (unsigned) addr >= 14 ) { #ifdef dprintf dprintf( "Wrote to I/O port %02X\n", (int) addr ); #endif } // envelope mode if ( addr == 13 ) { if ( !(data & 8) ) // convert modes 0-7 to proper equivalents data = (data & 4) ? 15 : 9; env.wave = env.modes [data - 7]; env.pos = -48; env.delay = 0; // will get set to envelope period in run_until() } regs [addr] = data; // handle period changes accurately int i = addr >> 1; if ( i < osc_count ) { blip_time_t period = (regs [i * 2 + 1] & 0x0F) * (0x100L * period_factor) + regs [i * 2] * period_factor; if ( !period ) period = period_factor; // adjust time of next timer expiration based on change in period osc_t& osc = oscs [i]; if ( (osc.delay += period - osc.period) < 0 ) osc.delay = 0; osc.period = period; } // TODO: same as above for envelope timer, and it also has a divide by two after it } int const noise_off = 0x08; int const tone_off = 0x01; void Ay_Apu::run_until( blip_time_t final_end_time ) { require( final_end_time >= last_time ); // noise period and initial values blip_time_t const noise_period_factor = period_factor * 2; // verified blip_time_t noise_period = (regs [6] & 0x1F) * noise_period_factor; if ( !noise_period ) noise_period = noise_period_factor; blip_time_t const old_noise_delay = noise.delay; blargg_ulong const old_noise_lfsr = noise.lfsr; // envelope period blip_time_t const env_period_factor = period_factor * 2; // verified blip_time_t env_period = (regs [12] * 0x100L + regs [11]) * env_period_factor; if ( !env_period ) env_period = env_period_factor; // same as period 1 on my AY chip if ( !env.delay ) env.delay = env_period; // run each osc separately for ( int index = 0; index < osc_count; index++ ) { osc_t* const osc = &oscs [index]; int osc_mode = regs [7] >> index; // output Blip_Buffer* const osc_output = osc->output; if ( !osc_output ) continue; osc_output->set_modified(); // period int half_vol = 0; blip_time_t inaudible_period = (blargg_ulong) (osc_output->clock_rate() + inaudible_freq) / (inaudible_freq * 2); if ( osc->period <= inaudible_period && !(osc_mode & tone_off) ) { half_vol = 1; // Actually around 60%, but 50% is close enough osc_mode |= tone_off; } // envelope blip_time_t start_time = last_time; blip_time_t end_time = final_end_time; int const vol_mode = regs [0x08 + index]; int volume = amp_table [vol_mode & 0x0F] >> half_vol; int osc_env_pos = env.pos; if ( vol_mode & 0x10 ) { volume = env.wave [osc_env_pos] >> half_vol; // use envelope only if it's a repeating wave or a ramp that hasn't finished if ( !(regs [13] & 1) || osc_env_pos < -32 ) { end_time = start_time + env.delay; if ( end_time >= final_end_time ) end_time = final_end_time; //if ( !(regs [12] | regs [11]) ) // dprintf( "Used envelope period 0\n" ); } else if ( !volume ) { osc_mode = noise_off | tone_off; } } else if ( !volume ) { osc_mode = noise_off | tone_off; } // tone time blip_time_t const period = osc->period; blip_time_t time = start_time + osc->delay; if ( osc_mode & tone_off ) // maintain tone's phase when off { blargg_long count = (final_end_time - time + period - 1) / period; time += count * period; osc->phase ^= count & 1; } // noise time blip_time_t ntime = final_end_time; blargg_ulong noise_lfsr = 1; if ( !(osc_mode & noise_off) ) { ntime = start_time + old_noise_delay; noise_lfsr = old_noise_lfsr; //if ( (regs [6] & 0x1F) == 0 ) // dprintf( "Used noise period 0\n" ); } // The following efficiently handles several cases (least demanding first): // * Tone, noise, and envelope disabled, where channel acts as 4-bit DAC // * Just tone or just noise, envelope disabled // * Envelope controlling tone and/or noise // * Tone and noise disabled, envelope enabled with high frequency // * Tone and noise together // * Tone and noise together with envelope // This loop only runs one iteration if envelope is disabled. If envelope // is being used as a waveform (tone and noise disabled), this loop will // still be reasonably efficient since the bulk of it will be skipped. while ( 1 ) { // current amplitude int amp = 0; if ( (osc_mode | osc->phase) & 1 & (osc_mode >> 3 | noise_lfsr) ) amp = volume; { int delta = amp - osc->last_amp; if ( delta ) { osc->last_amp = amp; synth_.offset( start_time, delta, osc_output ); } } // Run wave and noise interleved with each catching up to the other. // If one or both are disabled, their "current time" will be past end time, // so there will be no significant performance hit. if ( ntime < end_time || time < end_time ) { // Since amplitude was updated above, delta will always be +/- volume, // so we can avoid using last_amp every time to calculate the delta. int delta = amp * 2 - volume; int delta_non_zero = delta != 0; int phase = osc->phase | (osc_mode & tone_off); assert( tone_off == 0x01 ); do { // run noise blip_time_t end = end_time; if ( end_time > time ) end = time; if ( phase & delta_non_zero ) { while ( ntime <= end ) // must advance *past* time to avoid hang { int changed = noise_lfsr + 1; noise_lfsr = (-(noise_lfsr & 1) & 0x12000) ^ (noise_lfsr >> 1); if ( changed & 2 ) { delta = -delta; synth_.offset( ntime, delta, osc_output ); } ntime += noise_period; } } else { // 20 or more noise periods on average for some music blargg_long remain = end - ntime; blargg_long count = remain / noise_period; if ( remain >= 0 ) ntime += noise_period + count * noise_period; } // run tone end = end_time; if ( end_time > ntime ) end = ntime; if ( noise_lfsr & delta_non_zero ) { while ( time < end ) { delta = -delta; synth_.offset( time, delta, osc_output ); time += period; //phase ^= 1; } //assert( phase == (delta > 0) ); phase = unsigned (-delta) >> (CHAR_BIT * sizeof (unsigned) - 1); // (delta > 0) } else { // loop usually runs less than once //SUB_CASE_COUNTER( (time < end) * (end - time + period - 1) / period ); while ( time < end ) { time += period; phase ^= 1; } } } while ( time < end_time || ntime < end_time ); osc->last_amp = (delta + volume) >> 1; if ( !(osc_mode & tone_off) ) osc->phase = phase; } if ( end_time >= final_end_time ) break; // breaks first time when envelope is disabled // next envelope step if ( ++osc_env_pos >= 0 ) osc_env_pos -= 32; volume = env.wave [osc_env_pos] >> half_vol; start_time = end_time; end_time += env_period; if ( end_time > final_end_time ) end_time = final_end_time; } osc->delay = time - final_end_time; if ( !(osc_mode & noise_off) ) { noise.delay = ntime - final_end_time; noise.lfsr = noise_lfsr; } } // TODO: optimized saw wave envelope? // maintain envelope phase blip_time_t remain = final_end_time - last_time - env.delay; if ( remain >= 0 ) { blargg_long count = (remain + env_period) / env_period; env.pos += count; if ( env.pos >= 0 ) env.pos = (env.pos & 31) - 32; remain -= count * env_period; assert( -remain <= env_period ); } env.delay = -remain; assert( env.delay > 0 ); assert( env.pos < 0 ); last_time = final_end_time; }