pizza: cover your ears
This commit is contained in:
nattthebear
parent
dd9d1b2018
commit
04d4880564
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@ -27,7 +27,9 @@ namespace BizHawk.Emulation.Cores.Consoles.Nintendo.Gameboy
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public class FrameInfo
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{
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public IntPtr VideoBuffer;
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public IntPtr SoundBuffer;
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public int Clocks;
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public int Samples;
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public Buttons Keys;
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}
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@ -12,7 +12,7 @@ namespace BizHawk.Emulation.Cores.Consoles.Nintendo.Gameboy
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{
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[CoreAttributes("Pizza Boy", "Davide Berra", true, false, "c7bc6ee376028b3766de8d7a02e60ab794841f45",
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"https://github.com/davideberra/emu-pizza/", false)]
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public class Pizza : IEmulator, IInputPollable, IVideoProvider, IGameboyCommon
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public class Pizza : IEmulator, IInputPollable, IVideoProvider, IGameboyCommon, ISoundProvider
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{
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private LibPizza _pizza;
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private PeRunner _exe;
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@ -83,18 +83,21 @@ namespace BizHawk.Emulation.Cores.Consoles.Nintendo.Gameboy
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public unsafe void FrameAdvance(IController controller, bool render, bool rendersound = true)
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{
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fixed (int* vp = _videoBuffer)
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fixed (short* sp = _soundBuffer)
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{
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var targetClocks = TICKSPERFRAME - _tickOverflow;
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var frame = new LibPizza.FrameInfo
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{
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VideoBuffer = (IntPtr)vp,
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SoundBuffer = (IntPtr)sp,
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Clocks = targetClocks,
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Keys = GetButtons(controller)
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};
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_pizza.FrameAdvance(frame);
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_tickOverflow = frame.Clocks - targetClocks;
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_numSamples = frame.Samples;
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Frame++;
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}
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}
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@ -127,6 +130,40 @@ namespace BizHawk.Emulation.Cores.Consoles.Nintendo.Gameboy
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public bool DeterministicEmulation { get; private set; }
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public CoreComm CoreComm { get; }
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#region ISoundProvider
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private short[] _soundBuffer = new short[2048];
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private int _numSamples;
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public void SetSyncMode(SyncSoundMode mode)
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{
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if (mode == SyncSoundMode.Async)
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{
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throw new NotSupportedException("Async mode is not supported.");
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}
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}
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public void GetSamplesSync(out short[] samples, out int nsamp)
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{
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samples = _soundBuffer;
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nsamp = _numSamples;
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}
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public void GetSamplesAsync(short[] samples)
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{
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throw new InvalidOperationException("Async mode is not supported.");
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}
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public void DiscardSamples()
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{
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}
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public bool CanProvideAsync => false;
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public SyncSoundMode SyncMode => SyncSoundMode.Sync;
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#endregion
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#region IVideoProvider
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private int[] _videoBuffer = new int[160 * 144];
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@ -0,0 +1,344 @@
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/* blip_buf 1.1.0. http://www.slack.net/~ant/ */
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#include "blip_buf.h"
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#include <assert.h>
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#include <limits.h>
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#include <string.h>
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#include <stdlib.h>
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/* Library Copyright (C) 2003-2009 Shay Green. This library is free software;
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you can redistribute it and/or modify it under the terms of the GNU Lesser
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General Public License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version. This
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library is distributed in the hope that it will be useful, but WITHOUT ANY
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WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
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A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more
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details. You should have received a copy of the GNU Lesser General Public
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License along with this module; if not, write to the Free Software Foundation,
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Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */
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#if defined (BLARGG_TEST) && BLARGG_TEST
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#include "blargg_test.h"
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#endif
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/* Equivalent to ULONG_MAX >= 0xFFFFFFFF00000000.
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Avoids constants that don't fit in 32 bits. */
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#if ULONG_MAX/0xFFFFFFFF > 0xFFFFFFFF
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typedef unsigned long fixed_t;
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enum { pre_shift = 32 };
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#elif defined(ULLONG_MAX)
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typedef unsigned long long fixed_t;
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enum { pre_shift = 32 };
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#else
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typedef unsigned fixed_t;
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enum { pre_shift = 0 };
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#endif
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enum { time_bits = pre_shift + 20 };
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static fixed_t const time_unit = (fixed_t) 1 << time_bits;
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enum { bass_shift = 9 }; /* affects high-pass filter breakpoint frequency */
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enum { end_frame_extra = 2 }; /* allows deltas slightly after frame length */
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enum { half_width = 8 };
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enum { buf_extra = half_width*2 + end_frame_extra };
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enum { phase_bits = 5 };
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enum { phase_count = 1 << phase_bits };
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enum { delta_bits = 15 };
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enum { delta_unit = 1 << delta_bits };
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enum { frac_bits = time_bits - pre_shift };
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/* We could eliminate avail and encode whole samples in offset, but that would
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limit the total buffered samples to blip_max_frame. That could only be
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increased by decreasing time_bits, which would reduce resample ratio accuracy.
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*/
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/** Sample buffer that resamples to output rate and accumulates samples
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until they're read out */
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struct blip_t
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{
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fixed_t factor;
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fixed_t offset;
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int avail;
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int size;
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int integrator;
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};
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typedef int buf_t;
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/* probably not totally portable */
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#define SAMPLES( buf ) ((buf_t*) ((buf) + 1))
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/* Arithmetic (sign-preserving) right shift */
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#define ARITH_SHIFT( n, shift ) \
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((n) >> (shift))
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enum { max_sample = +32767 };
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enum { min_sample = -32768 };
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#define CLAMP( n ) \
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{\
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if ( (short) n != n )\
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n = ARITH_SHIFT( n, 16 ) ^ max_sample;\
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}
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static void check_assumptions( void )
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{
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int n;
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#if INT_MAX < 0x7FFFFFFF || UINT_MAX < 0xFFFFFFFF
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#error "int must be at least 32 bits"
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#endif
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assert( (-3 >> 1) == -2 ); /* right shift must preserve sign */
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n = max_sample * 2;
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CLAMP( n );
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assert( n == max_sample );
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n = min_sample * 2;
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CLAMP( n );
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assert( n == min_sample );
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assert( blip_max_ratio <= time_unit );
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assert( blip_max_frame <= (fixed_t) -1 >> time_bits );
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}
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blip_t* blip_new( int size )
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{
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blip_t* m;
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assert( size >= 0 );
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m = (blip_t*) malloc( sizeof *m + (size + buf_extra) * sizeof (buf_t) );
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if ( m )
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{
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m->factor = time_unit / blip_max_ratio;
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m->size = size;
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blip_clear( m );
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check_assumptions();
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}
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return m;
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}
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void blip_delete( blip_t* m )
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{
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if ( m != NULL )
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{
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/* Clear fields in case user tries to use after freeing */
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memset( m, 0, sizeof *m );
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free( m );
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}
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}
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void blip_set_rates( blip_t* m, double clock_rate, double sample_rate )
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{
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double factor = time_unit * sample_rate / clock_rate;
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m->factor = (fixed_t) factor;
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/* Fails if clock_rate exceeds maximum, relative to sample_rate */
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assert( 0 <= factor - m->factor && factor - m->factor < 1 );
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/* Avoid requiring math.h. Equivalent to
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m->factor = (int) ceil( factor ) */
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if ( m->factor < factor )
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m->factor++;
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/* At this point, factor is most likely rounded up, but could still
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have been rounded down in the floating-point calculation. */
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}
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void blip_clear( blip_t* m )
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{
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/* We could set offset to 0, factor/2, or factor-1. 0 is suitable if
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factor is rounded up. factor-1 is suitable if factor is rounded down.
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Since we don't know rounding direction, factor/2 accommodates either,
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with the slight loss of showing an error in half the time. Since for
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a 64-bit factor this is years, the halving isn't a problem. */
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m->offset = m->factor / 2;
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m->avail = 0;
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m->integrator = 0;
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memset( SAMPLES( m ), 0, (m->size + buf_extra) * sizeof (buf_t) );
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}
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int blip_clocks_needed( const blip_t* m, int samples )
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{
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fixed_t needed;
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/* Fails if buffer can't hold that many more samples */
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assert( samples >= 0 && m->avail + samples <= m->size );
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needed = (fixed_t) samples * time_unit;
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if ( needed < m->offset )
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return 0;
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return (needed - m->offset + m->factor - 1) / m->factor;
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}
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void blip_end_frame( blip_t* m, unsigned t )
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{
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fixed_t off = t * m->factor + m->offset;
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m->avail += off >> time_bits;
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m->offset = off & (time_unit - 1);
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/* Fails if buffer size was exceeded */
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assert( m->avail <= m->size );
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}
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int blip_samples_avail( const blip_t* m )
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{
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return m->avail;
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}
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static void remove_samples( blip_t* m, int count )
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{
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buf_t* buf = SAMPLES( m );
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int remain = m->avail + buf_extra - count;
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m->avail -= count;
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memmove( &buf [0], &buf [count], remain * sizeof buf [0] );
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memset( &buf [remain], 0, count * sizeof buf [0] );
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}
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int blip_read_samples( blip_t* m, short out [], int count, int stereo )
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{
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assert( count >= 0 );
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if ( count > m->avail )
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count = m->avail;
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if ( count )
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{
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int const step = stereo ? 2 : 1;
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buf_t const* in = SAMPLES( m );
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buf_t const* end = in + count;
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int sum = m->integrator;
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do
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{
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/* Eliminate fraction */
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int s = ARITH_SHIFT( sum, delta_bits );
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sum += *in++;
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CLAMP( s );
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*out = s;
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out += step;
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/* High-pass filter */
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sum -= s << (delta_bits - bass_shift);
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}
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while ( in != end );
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m->integrator = sum;
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remove_samples( m, count );
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}
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return count;
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}
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/* Things that didn't help performance on x86:
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__attribute__((aligned(128)))
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#define short int
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restrict
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*/
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/* Sinc_Generator( 0.9, 0.55, 4.5 ) */
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static short const bl_step [phase_count + 1] [half_width] =
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{
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{ 43, -115, 350, -488, 1136, -914, 5861,21022},
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{ 44, -118, 348, -473, 1076, -799, 5274,21001},
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{ 45, -121, 344, -454, 1011, -677, 4706,20936},
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{ 46, -122, 336, -431, 942, -549, 4156,20829},
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{ 47, -123, 327, -404, 868, -418, 3629,20679},
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{ 47, -122, 316, -375, 792, -285, 3124,20488},
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{ 47, -120, 303, -344, 714, -151, 2644,20256},
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{ 46, -117, 289, -310, 634, -17, 2188,19985},
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{ 46, -114, 273, -275, 553, 117, 1758,19675},
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{ 44, -108, 255, -237, 471, 247, 1356,19327},
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{ 43, -103, 237, -199, 390, 373, 981,18944},
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{ 42, -98, 218, -160, 310, 495, 633,18527},
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{ 40, -91, 198, -121, 231, 611, 314,18078},
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{ 38, -84, 178, -81, 153, 722, 22,17599},
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{ 36, -76, 157, -43, 80, 824, -241,17092},
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{ 34, -68, 135, -3, 8, 919, -476,16558},
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{ 32, -61, 115, 34, -60, 1006, -683,16001},
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{ 29, -52, 94, 70, -123, 1083, -862,15422},
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{ 27, -44, 73, 106, -184, 1152,-1015,14824},
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{ 25, -36, 53, 139, -239, 1211,-1142,14210},
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{ 22, -27, 34, 170, -290, 1261,-1244,13582},
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{ 20, -20, 16, 199, -335, 1301,-1322,12942},
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{ 18, -12, -3, 226, -375, 1331,-1376,12293},
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{ 15, -4, -19, 250, -410, 1351,-1408,11638},
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{ 13, 3, -35, 272, -439, 1361,-1419,10979},
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{ 11, 9, -49, 292, -464, 1362,-1410,10319},
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{ 9, 16, -63, 309, -483, 1354,-1383, 9660},
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{ 7, 22, -75, 322, -496, 1337,-1339, 9005},
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{ 6, 26, -85, 333, -504, 1312,-1280, 8355},
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{ 4, 31, -94, 341, -507, 1278,-1205, 7713},
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{ 3, 35, -102, 347, -506, 1238,-1119, 7082},
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{ 1, 40, -110, 350, -499, 1190,-1021, 6464},
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{ 0, 43, -115, 350, -488, 1136, -914, 5861}
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};
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/* Shifting by pre_shift allows calculation using unsigned int rather than
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possibly-wider fixed_t. On 32-bit platforms, this is likely more efficient.
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And by having pre_shift 32, a 32-bit platform can easily do the shift by
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simply ignoring the low half. */
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void blip_add_delta( blip_t* m, unsigned time, int delta )
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{
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unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
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buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
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int const phase_shift = frac_bits - phase_bits;
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int phase = fixed >> phase_shift & (phase_count - 1);
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short const* in = bl_step [phase];
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short const* rev = bl_step [phase_count - phase];
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int interp = fixed >> (phase_shift - delta_bits) & (delta_unit - 1);
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int delta2 = (delta * interp) >> delta_bits;
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delta -= delta2;
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/* Fails if buffer size was exceeded */
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assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
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out [0] += in[0]*delta + in[half_width+0]*delta2;
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out [1] += in[1]*delta + in[half_width+1]*delta2;
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out [2] += in[2]*delta + in[half_width+2]*delta2;
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out [3] += in[3]*delta + in[half_width+3]*delta2;
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out [4] += in[4]*delta + in[half_width+4]*delta2;
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out [5] += in[5]*delta + in[half_width+5]*delta2;
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out [6] += in[6]*delta + in[half_width+6]*delta2;
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out [7] += in[7]*delta + in[half_width+7]*delta2;
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in = rev;
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out [ 8] += in[7]*delta + in[7-half_width]*delta2;
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out [ 9] += in[6]*delta + in[6-half_width]*delta2;
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out [10] += in[5]*delta + in[5-half_width]*delta2;
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out [11] += in[4]*delta + in[4-half_width]*delta2;
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out [12] += in[3]*delta + in[3-half_width]*delta2;
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out [13] += in[2]*delta + in[2-half_width]*delta2;
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out [14] += in[1]*delta + in[1-half_width]*delta2;
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out [15] += in[0]*delta + in[0-half_width]*delta2;
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}
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void blip_add_delta_fast( blip_t* m, unsigned time, int delta )
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{
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unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
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buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
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int interp = fixed >> (frac_bits - delta_bits) & (delta_unit - 1);
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int delta2 = delta * interp;
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/* Fails if buffer size was exceeded */
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assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
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out [7] += delta * delta_unit - delta2;
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out [8] += delta2;
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}
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@ -0,0 +1,72 @@
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/** \file
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Sample buffer that resamples from input clock rate to output sample rate */
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/* blip_buf 1.1.0 */
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#ifndef BLIP_BUF_H
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#define BLIP_BUF_H
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#ifdef __cplusplus
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extern "C" {
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#endif
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/** First parameter of most functions is blip_t*, or const blip_t* if nothing
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is changed. */
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typedef struct blip_t blip_t;
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/** Creates new buffer that can hold at most sample_count samples. Sets rates
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so that there are blip_max_ratio clocks per sample. Returns pointer to new
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buffer, or NULL if insufficient memory. */
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blip_t* blip_new( int sample_count );
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/** Sets approximate input clock rate and output sample rate. For every
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clock_rate input clocks, approximately sample_rate samples are generated. */
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void blip_set_rates( blip_t*, double clock_rate, double sample_rate );
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||||
|
||||
enum { /** Maximum clock_rate/sample_rate ratio. For a given sample_rate,
|
||||
clock_rate must not be greater than sample_rate*blip_max_ratio. */
|
||||
blip_max_ratio = 1 << 20 };
|
||||
|
||||
/** Clears entire buffer. Afterwards, blip_samples_avail() == 0. */
|
||||
void blip_clear( blip_t* );
|
||||
|
||||
/** Adds positive/negative delta into buffer at specified clock time. */
|
||||
void blip_add_delta( blip_t*, unsigned int clock_time, int delta );
|
||||
|
||||
/** Same as blip_add_delta(), but uses faster, lower-quality synthesis. */
|
||||
void blip_add_delta_fast( blip_t*, unsigned int clock_time, int delta );
|
||||
|
||||
/** Length of time frame, in clocks, needed to make sample_count additional
|
||||
samples available. */
|
||||
int blip_clocks_needed( const blip_t*, int sample_count );
|
||||
|
||||
enum { /** Maximum number of samples that can be generated from one time frame. */
|
||||
blip_max_frame = 4000 };
|
||||
|
||||
/** Makes input clocks before clock_duration available for reading as output
|
||||
samples. Also begins new time frame at clock_duration, so that clock time 0 in
|
||||
the new time frame specifies the same clock as clock_duration in the old time
|
||||
frame specified. Deltas can have been added slightly past clock_duration (up to
|
||||
however many clocks there are in two output samples). */
|
||||
void blip_end_frame( blip_t*, unsigned int clock_duration );
|
||||
|
||||
/** Number of buffered samples available for reading. */
|
||||
int blip_samples_avail( const blip_t* );
|
||||
|
||||
/** Reads and removes at most 'count' samples and writes them to 'out'. If
|
||||
'stereo' is true, writes output to every other element of 'out', allowing easy
|
||||
interleaving of two buffers into a stereo sample stream. Outputs 16-bit signed
|
||||
samples. Returns number of samples actually read. */
|
||||
int blip_read_samples( blip_t*, short out [], int count, int stereo );
|
||||
|
||||
/** Frees buffer. No effect if NULL is passed. */
|
||||
void blip_delete( blip_t* );
|
||||
|
||||
|
||||
/* Deprecated */
|
||||
typedef blip_t blip_buffer_t;
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
||||
|
|
@ -222,10 +222,6 @@ void cycles_step()
|
|||
/* channel four */
|
||||
if (sound.channel_four.cycles_next == cycles.cnt)
|
||||
sound_step_ch4();
|
||||
|
||||
/* time to generate a sample? */
|
||||
if (sound.sample_cycles_next_rounded == cycles.cnt)
|
||||
sound_step_sample();
|
||||
|
||||
/* update timer state */
|
||||
if (cycles.cnt == timer.next)
|
||||
|
|
|
|||
|
|
@ -39,9 +39,6 @@ char gameboy_inited = 0;
|
|||
|
||||
void gameboy_init()
|
||||
{
|
||||
/* init global values */
|
||||
// global_init();
|
||||
|
||||
/* init z80 */
|
||||
z80_init();
|
||||
|
||||
|
|
|
|||
|
|
@ -23,6 +23,7 @@
|
|||
#include "mmu.h"
|
||||
#include "sound.h"
|
||||
#include "utils.h"
|
||||
#include "sound_output.h"
|
||||
|
||||
#include <string.h>
|
||||
#include <strings.h>
|
||||
|
|
@ -31,26 +32,87 @@
|
|||
/* super variable for audio controller */
|
||||
sound_t sound;
|
||||
|
||||
/* global for output frequency */
|
||||
int sound_output_rate = 48000;
|
||||
int sound_output_rate_fifth = 48; // 48000 / 5;
|
||||
// functions to connect to blip buf
|
||||
static int16_t last_sample[8];
|
||||
|
||||
#define BLIP_LEFT(v,i) if(1){int32_t d = (v)-last_sample[(i)*2];blip_left(d);last_sample[(i)*2] = (v);}
|
||||
#define BLIP_RIGHT(v,i) if(1){int32_t d = (v)-last_sample[(i)*2+1];blip_right(d);last_sample[(i)*2+1] = (v);}
|
||||
|
||||
static void blip_ch1()
|
||||
{
|
||||
if (sound.channel_one.active)
|
||||
{
|
||||
if (sound.nr51->ch1_to_so1)
|
||||
BLIP_RIGHT(sound.channel_one.sample, 0);
|
||||
if (sound.nr51->ch1_to_so2)
|
||||
BLIP_LEFT(sound.channel_one.sample, 0);
|
||||
}
|
||||
}
|
||||
static void blip_ch2()
|
||||
{
|
||||
if (sound.channel_two.active)
|
||||
{
|
||||
if (sound.nr51->ch2_to_so1)
|
||||
BLIP_RIGHT(sound.channel_two.sample, 1);
|
||||
if (sound.nr51->ch2_to_so2)
|
||||
BLIP_LEFT(sound.channel_two.sample, 1);
|
||||
}
|
||||
}
|
||||
static void blip_ch3()
|
||||
{
|
||||
if (sound.channel_three.active)
|
||||
{
|
||||
uint16_t sample;
|
||||
uint8_t shift = (sound.nr32->volume_code == 0 ?
|
||||
4 : sound.nr32->volume_code - 1);
|
||||
|
||||
/* volume is zero in any case */
|
||||
if (shift == 4)
|
||||
sample = 0;
|
||||
else
|
||||
{
|
||||
/* apply volume change */
|
||||
uint8_t idx = sound.channel_three.index;
|
||||
uint16_t s;
|
||||
|
||||
/* extract current sample */
|
||||
if ((idx & 0x01) == 0)
|
||||
s = (sound.wave_table[idx >> 1] & 0xf0) >> 4;
|
||||
else
|
||||
s = sound.wave_table[idx >> 1] & 0x0f;
|
||||
|
||||
/* transform it into signed 16 bit sample */
|
||||
sample = ((s * 0x222) >> shift);
|
||||
}
|
||||
|
||||
|
||||
/* not silence? */
|
||||
if (sample != 0)
|
||||
{
|
||||
if (sound.nr51->ch3_to_so1)
|
||||
BLIP_RIGHT(sample, 2);
|
||||
if (sound.nr51->ch3_to_so2)
|
||||
BLIP_LEFT(sample, 2);
|
||||
}
|
||||
}
|
||||
}
|
||||
static void blip_ch4()
|
||||
{
|
||||
if (sound.channel_four.active)
|
||||
{
|
||||
if (sound.nr51->ch4_to_so1)
|
||||
BLIP_RIGHT(sound.channel_four.sample, 3);
|
||||
if (sound.nr51->ch4_to_so2)
|
||||
BLIP_LEFT(sound.channel_four.sample, 3);
|
||||
}
|
||||
}
|
||||
|
||||
/* internal prototypes */
|
||||
size_t sound_available_samples();
|
||||
void sound_envelope_step();
|
||||
void sound_length_ctrl_step();
|
||||
void sound_push_samples(int16_t l, int16_t r);
|
||||
void sound_push_sample(int16_t s);
|
||||
void sound_read_samples(int len, int16_t *buf);
|
||||
void sound_rebuild_wave();
|
||||
void sound_sweep_step();
|
||||
void sound_write_wave(uint16_t a, uint8_t v);
|
||||
|
||||
int sound_get_samples()
|
||||
{
|
||||
return SOUND_SAMPLES; // sound_output_rate / 10;
|
||||
}
|
||||
|
||||
void sound_init_pointers()
|
||||
{
|
||||
/* point sound structures to their memory areas */
|
||||
|
|
@ -92,46 +154,15 @@ void sound_init()
|
|||
/* point sound structures to their memory areas */
|
||||
sound_init_pointers();
|
||||
|
||||
/* steps length */
|
||||
sound.step_int = 4;
|
||||
sound.step_int1000 = 4000;
|
||||
|
||||
/* buffer stuff */
|
||||
sound.buf_wr = 0;
|
||||
sound.buf_rd = 0;
|
||||
sound.buf_available = 0;
|
||||
|
||||
/* how many cpu cycles we need to emit a 512hz clock (frame sequencer) */
|
||||
sound.fs_cycles = 4194304 / 512;
|
||||
|
||||
/* how many cpu cycles to generate a single frame seq clock? */
|
||||
sound.fs_cycles_next = sound.fs_cycles;
|
||||
|
||||
/* how many cpu cycles to generate a sample */
|
||||
sound.sample_cycles = (uint_fast32_t) (((double) 4194304 /
|
||||
(double) sound_output_rate) * 1000);
|
||||
|
||||
sound.sample_cycles_next = sound.sample_cycles / 1000;
|
||||
sound.sample_cycles_next_rounded = sound.sample_cycles_next & 0xFFFFFFFC;
|
||||
|
||||
/* no, i'm not empty */
|
||||
sound.buf_empty = 0;
|
||||
}
|
||||
|
||||
void sound_set_speed(char dbl)
|
||||
{
|
||||
return;
|
||||
|
||||
if (dbl)
|
||||
{
|
||||
sound.step_int = 2;
|
||||
sound.step_int1000 = 2000;
|
||||
}
|
||||
else
|
||||
{
|
||||
sound.step_int = 4;
|
||||
sound.step_int1000 = 4000;
|
||||
}
|
||||
}
|
||||
|
||||
/* update sound internal state given CPU T-states */
|
||||
|
|
@ -155,6 +186,10 @@ void sound_step_fs()
|
|||
/* envelope works at 64hz */
|
||||
if (sound.fs_cycles_idx == 7)
|
||||
sound_envelope_step();
|
||||
blip_ch1();
|
||||
blip_ch2();
|
||||
blip_ch3();
|
||||
blip_ch4();
|
||||
}
|
||||
|
||||
/* update all channels */
|
||||
|
|
@ -172,6 +207,7 @@ void sound_step_ch1()
|
|||
|
||||
/* go back */
|
||||
sound.channel_one.duty_cycles_next += sound.channel_one.duty_cycles;
|
||||
blip_ch1();
|
||||
}
|
||||
|
||||
void sound_step_ch2()
|
||||
|
|
@ -188,6 +224,7 @@ void sound_step_ch2()
|
|||
|
||||
/* go back */
|
||||
sound.channel_two.duty_cycles_next += sound.channel_two.duty_cycles;
|
||||
blip_ch2();
|
||||
}
|
||||
|
||||
void sound_step_ch3()
|
||||
|
|
@ -206,6 +243,7 @@ void sound_step_ch3()
|
|||
/* qty of cpu ticks needed for a wave sample change */
|
||||
sound.channel_three.cycles = ((2048 - freq) * 2) << global_cpu_double_speed;
|
||||
sound.channel_three.cycles_next += sound.channel_three.cycles;
|
||||
blip_ch3();
|
||||
}
|
||||
|
||||
void sound_step_ch4()
|
||||
|
|
@ -236,109 +274,8 @@ void sound_step_ch4()
|
|||
sound.channel_four.sample = sound.channel_four.volume;
|
||||
|
||||
/* qty of cpu ticks needed for a wave sample change */
|
||||
sound.channel_four.cycles_next += sound.channel_four.period_lfsr;
|
||||
}
|
||||
|
||||
void sound_step_sample()
|
||||
{
|
||||
uint_fast32_t zum = sound.sample_cycles + sound.sample_cycles_remainder;
|
||||
|
||||
sound.sample_cycles_next += ((zum / 1000) << global_cpu_double_speed);
|
||||
sound.sample_cycles_next_rounded =
|
||||
sound.sample_cycles_next & 0xFFFFFFFC;
|
||||
sound.sample_cycles_remainder = zum % 1000;
|
||||
|
||||
/* update output frame counter */
|
||||
sound.frame_counter++;
|
||||
|
||||
/* DAC turned off? */
|
||||
if (sound.nr30->dac == 0 &&
|
||||
sound.channel_one.active == 0 &&
|
||||
sound.channel_two.active == 0 &&
|
||||
sound.channel_four.active == 0)
|
||||
{
|
||||
sound_push_samples(0, 0);
|
||||
return;
|
||||
}
|
||||
|
||||
int16_t sample_left = 0;
|
||||
int16_t sample_right = 0;
|
||||
int16_t sample = 0;
|
||||
|
||||
/* time to generate a sample! sum all the fields */
|
||||
if (sound.channel_one.active) // && sound.channel_one.sample)
|
||||
{
|
||||
/* to the right? */
|
||||
if (sound.nr51->ch1_to_so1)
|
||||
sample_right += sound.channel_one.sample;
|
||||
|
||||
/* to the left? */
|
||||
if (sound.nr51->ch1_to_so2)
|
||||
sample_left += sound.channel_one.sample;
|
||||
}
|
||||
|
||||
if (sound.channel_two.active) // && sound.channel_two.sample)
|
||||
{
|
||||
/* to the right? */
|
||||
if (sound.nr51->ch2_to_so1)
|
||||
sample_right += sound.channel_two.sample;
|
||||
|
||||
/* to the left? */
|
||||
if (sound.nr51->ch2_to_so2)
|
||||
sample_left += sound.channel_two.sample;
|
||||
}
|
||||
|
||||
if (sound.channel_three.active)
|
||||
{
|
||||
uint8_t shift = (sound.nr32->volume_code == 0 ?
|
||||
4 : sound.nr32->volume_code - 1);
|
||||
|
||||
/* volume is zero in any case */
|
||||
if (shift == 4)
|
||||
sample = 0;
|
||||
else
|
||||
{
|
||||
/* apply volume change */
|
||||
uint8_t idx = sound.channel_three.index;
|
||||
uint16_t s;
|
||||
|
||||
/* extract current sample */
|
||||
if ((idx & 0x01) == 0)
|
||||
s = (sound.wave_table[idx >> 1] & 0xf0) >> 4;
|
||||
else
|
||||
s = sound.wave_table[idx >> 1] & 0x0f;
|
||||
|
||||
/* transform it into signed 16 bit sample */
|
||||
sample = ((s * 0x222) >> shift);
|
||||
}
|
||||
|
||||
|
||||
/* not silence? */
|
||||
if (sample != 0)
|
||||
{
|
||||
/* to the right? */
|
||||
if (sound.nr51->ch3_to_so1)
|
||||
sample_right += sample;
|
||||
|
||||
/* to the left? */
|
||||
if (sound.nr51->ch3_to_so2)
|
||||
sample_left += sample;
|
||||
}
|
||||
}
|
||||
|
||||
if (sound.channel_four.active)
|
||||
{
|
||||
/* to the right? */
|
||||
if (sound.nr51->ch4_to_so1)
|
||||
sample_right += sound.channel_four.sample;
|
||||
|
||||
/* to the left? */
|
||||
if (sound.nr51->ch4_to_so2)
|
||||
sample_left += sound.channel_four.sample;
|
||||
}
|
||||
|
||||
/* push the sum of all channels samples */
|
||||
sound_push_samples(sample_left, sample_right);
|
||||
sound.channel_four.cycles_next += sound.channel_four.period_lfsr;
|
||||
blip_ch4();
|
||||
}
|
||||
|
||||
/* update length of channel1 */
|
||||
|
|
@ -377,116 +314,6 @@ void sound_length_ctrl_step()
|
|||
&sound.channel_four.active);
|
||||
}
|
||||
|
||||
void sound_read_buffer(void *userdata, uint8_t *stream, int snd_len)
|
||||
{
|
||||
/* requester snd_len is expressed in byte, */
|
||||
/* so divided by to to obtain wanted 16 bits samples */
|
||||
sound_read_samples(snd_len / 2, (int16_t *) stream);
|
||||
}
|
||||
|
||||
void sound_push_samples(int16_t l, int16_t r)
|
||||
{
|
||||
/* store them in tmp buffer */
|
||||
sound.buf_tmp[sound.buf_tmp_wr++] = l;
|
||||
sound.buf_tmp[sound.buf_tmp_wr++] = r;
|
||||
|
||||
if (sound.buf_tmp_wr == SOUND_BUF_TMP_SZ)
|
||||
{
|
||||
unsigned int i;
|
||||
|
||||
/* put them in circular shared buffer */
|
||||
for (i=0; i<SOUND_BUF_TMP_SZ; i++)
|
||||
sound_push_sample(sound.buf_tmp[i]);
|
||||
|
||||
/* reset counter */
|
||||
sound.buf_tmp_wr = 0;
|
||||
}
|
||||
}
|
||||
|
||||
/* push a single sample data into circular buffer */
|
||||
void sound_push_sample(int16_t s)
|
||||
{
|
||||
/* assign sample value */
|
||||
sound.buf[sound.buf_wr] = s;
|
||||
|
||||
/* update write index */
|
||||
sound.buf_wr++;
|
||||
|
||||
if (sound.buf_wr == SOUND_BUF_SZ)
|
||||
sound.buf_wr = 0;
|
||||
|
||||
/* update available samples */
|
||||
sound.buf_available++;
|
||||
|
||||
/* if it's locked and we got enough samples, unlock */
|
||||
if (sound.buf_empty && sound.buf_available == (SOUND_SAMPLES * 2))
|
||||
{
|
||||
sound.buf_empty = 0;
|
||||
|
||||
//pthread_cond_signal(&sound_cond);
|
||||
}
|
||||
|
||||
/* wait for the audio to be played */
|
||||
if (sound.buf_available == SOUND_BUF_SZ)
|
||||
{
|
||||
/* if full, just discard an old sample */
|
||||
sound.buf_available--;
|
||||
|
||||
/* step forward for rd pointer */
|
||||
sound.buf_rd++;
|
||||
|
||||
if (sound.buf_rd == SOUND_BUF_SZ)
|
||||
sound.buf_rd = 0;
|
||||
}
|
||||
}
|
||||
|
||||
/* calculate the available samples in circula buffer */
|
||||
size_t sound_available_samples()
|
||||
{
|
||||
if (sound.buf_rd > sound.buf_wr)
|
||||
return sound.buf_wr + SOUND_BUF_SZ - sound.buf_rd;
|
||||
|
||||
return sound.buf_wr - sound.buf_rd;
|
||||
}
|
||||
|
||||
/* read a block of data from circular buffer */
|
||||
void sound_read_samples(int to_read, int16_t *buf)
|
||||
{
|
||||
/* not enough samples? read what we got */
|
||||
if (sound.buf_available < to_read)
|
||||
{
|
||||
/* stop until we got enough samples */
|
||||
sound.buf_empty = 1;
|
||||
|
||||
//while (sound.buf_empty && !global_quit)
|
||||
//pthread_cond_wait(&sound_cond, &sound_mutex);
|
||||
}
|
||||
|
||||
if (sound.buf_rd + to_read >= SOUND_BUF_SZ)
|
||||
{
|
||||
/* overlaps the end of the buffer? copy in 2 phases */
|
||||
size_t first_block = SOUND_BUF_SZ - sound.buf_rd;
|
||||
|
||||
memcpy(buf, &sound.buf[sound.buf_rd], first_block * 2);
|
||||
|
||||
memcpy(&buf[first_block], sound.buf, (to_read - first_block) * 2);
|
||||
|
||||
/* set the new read index */
|
||||
sound.buf_rd = to_read - first_block;
|
||||
}
|
||||
else
|
||||
{
|
||||
/* a single memcpy is enough */
|
||||
memcpy(buf, &sound.buf[sound.buf_rd], to_read * 2);
|
||||
|
||||
/* update read index */
|
||||
sound.buf_rd += to_read;
|
||||
}
|
||||
|
||||
/* update avaiable samples */
|
||||
sound.buf_available -= to_read;
|
||||
}
|
||||
|
||||
/* calc the new frequency by sweep module */
|
||||
uint_fast32_t sound_sweep_calc()
|
||||
{
|
||||
|
|
@ -745,21 +572,6 @@ uint8_t sound_read_reg(uint16_t a, uint8_t v)
|
|||
}
|
||||
}
|
||||
|
||||
void sound_set_output_rate(int freq)
|
||||
{
|
||||
sound_output_rate = freq;
|
||||
sound_output_rate_fifth = freq / 5;
|
||||
|
||||
double cpu_base_freq = 4194304;
|
||||
|
||||
/* calc cycles needed to generate a sample */
|
||||
sound.sample_cycles = (uint_fast32_t) (((double) cpu_base_freq /
|
||||
(double) sound_output_rate) * 1000);
|
||||
|
||||
sound.sample_cycles_next = sound.sample_cycles / 1000;
|
||||
sound.sample_cycles_next_rounded = sound.sample_cycles_next & 0xFFFFFFFC;
|
||||
}
|
||||
|
||||
void sound_write_reg(uint16_t a, uint8_t v)
|
||||
{
|
||||
/* when turned off, only write to NR52 (0xFF26) is legit */
|
||||
|
|
@ -1351,57 +1163,3 @@ void sound_write_wave(uint16_t a, uint8_t v)
|
|||
|
||||
sound.wave_table[a - 0xFF30] = v;
|
||||
}
|
||||
|
||||
void sound_rebuild_wave()
|
||||
{
|
||||
int sample;
|
||||
uint8_t shift = (sound.nr32->volume_code == 0 ?
|
||||
4 : sound.nr32->volume_code - 1);
|
||||
int i;
|
||||
|
||||
int min = 0;
|
||||
int max = 0;
|
||||
|
||||
/* fill wave buffer */
|
||||
for (i=0; i<16; i++)
|
||||
{
|
||||
/* read higher nibble */
|
||||
sample = (sound.wave_table[i] & 0xf0) >> 4;
|
||||
|
||||
/* apply volume change */
|
||||
sample >>= shift;
|
||||
|
||||
/* convert to signed int 16 */
|
||||
sample = (sample * 0x1111) - 0x8000;
|
||||
|
||||
if (sample < min)
|
||||
min = sample;
|
||||
if (sample > max)
|
||||
max = sample;
|
||||
|
||||
/* save it into rendered wave table */
|
||||
sound.channel_three.wave[i * 2] = (int16_t) sample;
|
||||
|
||||
/* do the same with lowest nibble */
|
||||
sample = (sound.wave_table[i] & 0x0f);
|
||||
|
||||
/* apply volume change */
|
||||
sample >>= shift;
|
||||
|
||||
/* convert to signed int 16 */
|
||||
sample = (sample * 0x1111) - 0x8000;
|
||||
|
||||
/* save it into rendered wave table */
|
||||
sound.channel_three.wave[(i * 2) + 1] = (int16_t) sample;
|
||||
|
||||
if (sample < min)
|
||||
min = sample;
|
||||
if (sample > max)
|
||||
max = sample;
|
||||
}
|
||||
|
||||
/* set the new current sample */
|
||||
sound.channel_three.sample =
|
||||
sound.channel_three.wave[sound.channel_three.index];
|
||||
|
||||
}
|
||||
|
|
|
|||
|
|
@ -20,10 +20,7 @@
|
|||
#ifndef __SOUND_HDR__
|
||||
#define __SOUND_HDR__
|
||||
|
||||
#define SOUND_FREQ_MAX 48000
|
||||
#define SOUND_SAMPLES 4096
|
||||
#define SOUND_BUF_SZ (SOUND_SAMPLES * 3)
|
||||
#define SOUND_BUF_TMP_SZ (SOUND_SAMPLES / 2)
|
||||
|
||||
typedef struct nr10_s
|
||||
{
|
||||
|
|
@ -278,16 +275,6 @@ typedef struct sound_s
|
|||
/* emulation speed stuff */
|
||||
uint_fast16_t frame_counter;
|
||||
|
||||
/* circular audio buffer stuff */
|
||||
uint_fast16_t buf_rd;
|
||||
uint_fast16_t buf_wr;
|
||||
uint_fast16_t buf_available;
|
||||
uint_fast16_t buf_empty;
|
||||
uint_fast16_t buf_full;
|
||||
int16_t buf[SOUND_BUF_SZ];
|
||||
int16_t buf_tmp[SOUND_BUF_TMP_SZ];
|
||||
uint_fast16_t buf_tmp_wr;
|
||||
|
||||
/* output rate */
|
||||
uint_fast32_t output_rate;
|
||||
|
||||
|
|
@ -295,36 +282,19 @@ typedef struct sound_s
|
|||
uint_fast32_t fs_cycles;
|
||||
uint_fast32_t fs_cycles_idx;
|
||||
uint64_t fs_cycles_next;
|
||||
uint_fast32_t sample_cycles;
|
||||
uint_fast32_t sample_cycles_remainder;
|
||||
uint_fast32_t sample_cycles_next;
|
||||
uint64_t sample_cycles_next_rounded;
|
||||
|
||||
/* steps length */
|
||||
uint_fast32_t step_int;
|
||||
uint_fast32_t step_int1000;
|
||||
|
||||
uint_fast32_t spare;
|
||||
uint_fast32_t spare2;
|
||||
|
||||
} sound_t;
|
||||
|
||||
extern sound_t sound;
|
||||
|
||||
/* prototypes */
|
||||
void sound_change_emulation_speed();
|
||||
int sound_get_samples();
|
||||
void sound_init();
|
||||
void sound_read_buffer(void *userdata, uint8_t *stream, int snd_len);
|
||||
uint8_t sound_read_reg(uint16_t a, uint8_t v);
|
||||
void sound_set_speed(char dbl);
|
||||
void sound_set_output_rate(int freq);
|
||||
void sound_step_fs();
|
||||
void sound_step_ch1();
|
||||
void sound_step_ch2();
|
||||
void sound_step_ch3();
|
||||
void sound_step_ch4();
|
||||
void sound_step_sample();
|
||||
void sound_write_reg(uint16_t a, uint8_t v);
|
||||
|
||||
#endif
|
||||
|
|
|
|||
|
|
@ -0,0 +1,37 @@
|
|||
#include "../blip_buf/blip_buf.h"
|
||||
#include "sound_output.h"
|
||||
#include "cycles.h"
|
||||
|
||||
static blip_t* lb;
|
||||
static blip_t* rb;
|
||||
static uint64_t startclock;
|
||||
|
||||
#define RELATIVECLOCK (cycles.sampleclock - startclock)
|
||||
|
||||
void blip_left(int delta)
|
||||
{
|
||||
if (delta)
|
||||
blip_add_delta(lb, RELATIVECLOCK, delta);
|
||||
}
|
||||
void blip_right(int delta)
|
||||
{
|
||||
if (delta)
|
||||
blip_add_delta(rb, RELATIVECLOCK, delta);
|
||||
}
|
||||
|
||||
void sound_output_init(double clock_rate, double sample_rate)
|
||||
{
|
||||
lb = blip_new(1024);
|
||||
rb = blip_new(1024);
|
||||
blip_set_rates(lb, clock_rate, sample_rate);
|
||||
blip_set_rates(rb, clock_rate, sample_rate);
|
||||
}
|
||||
int sound_output_read(int16_t* output)
|
||||
{
|
||||
blip_end_frame(lb, RELATIVECLOCK);
|
||||
blip_end_frame(rb, RELATIVECLOCK);
|
||||
startclock = cycles.sampleclock;
|
||||
int ret = blip_read_samples(lb, output, 2048, 1);
|
||||
blip_read_samples(rb, output + 1, 2048, 1);
|
||||
return ret;
|
||||
}
|
||||
|
|
@ -0,0 +1,9 @@
|
|||
#pragma once
|
||||
|
||||
#include <stdint.h>
|
||||
|
||||
void blip_left(int delta);
|
||||
void blip_right(int delta);
|
||||
|
||||
void sound_output_init(double clock_rate, double sample_rate);
|
||||
int sound_output_read(int16_t* output);
|
||||
|
|
@ -35,6 +35,7 @@
|
|||
#include "serial.h"
|
||||
#include "utils.h"
|
||||
#include "mmu.h"
|
||||
#include "sound_output.h"
|
||||
|
||||
/* proto */
|
||||
void cb();
|
||||
|
|
@ -74,15 +75,14 @@ EXPORT int Init(const void *rom, int romlen)
|
|||
/* init GPU */
|
||||
gpu_init(&cb);
|
||||
|
||||
/* set sound output rate */
|
||||
sound_set_output_rate(44100);
|
||||
|
||||
/* set rumble cb */
|
||||
mmu_set_rumble_cb(&rumble_cb);
|
||||
|
||||
/* get frame buffer reference */
|
||||
fb = gpu_get_frame_buffer();
|
||||
|
||||
sound_output_init(2 * 1024 * 1024, 44100);
|
||||
|
||||
return 1;
|
||||
}
|
||||
|
||||
|
|
@ -91,7 +91,9 @@ static uint32_t fb32[160 * 144];
|
|||
typedef struct
|
||||
{
|
||||
uint32_t* vbuff;
|
||||
int16_t* sbuff;
|
||||
int32_t clocks; // desired(in) actual(out) time to run; 2MHZ
|
||||
int32_t samples; // actual number of samples produced
|
||||
uint16_t keys; // keypad input
|
||||
} frameinfo_t;
|
||||
|
||||
|
|
@ -103,6 +105,7 @@ EXPORT void FrameAdvance(frameinfo_t* frame)
|
|||
gameboy_run(current + frame->clocks);
|
||||
frame->clocks = cycles.sampleclock - current;
|
||||
memcpy(frame->vbuff, fb32, 160 * 144 * sizeof(uint32_t));
|
||||
frame->samples = sound_output_read(frame->sbuff);
|
||||
}
|
||||
|
||||
EXPORT int IsCGB(void)
|
||||
|
|
|
|||
Loading…
Reference in New Issue