mirror of
https://github.com/tildearrow/furnace.git
synced 2024-10-31 18:12:40 +00:00
earliest playback
no sound, just terminal output
This commit is contained in:
tildearrow
parent
9d17655836
commit
f810fc0c3c
15 changed files with 901 additions and 22 deletions
|
|
@ -21,7 +21,14 @@ if (HAVE_JACK)
|
|||
list(APPEND AUDIO_SOURCES src/audio/jack.cpp)
|
||||
endif()
|
||||
|
||||
set(ENGINE_SOURCES src/log.cpp src/engine/safeReader.cpp src/engine/engine.cpp)
|
||||
set(ENGINE_SOURCES
|
||||
src/log.cpp
|
||||
src/engine/blip_buf.c
|
||||
src/engine/safeReader.cpp
|
||||
src/engine/engine.cpp
|
||||
src/engine/playback.cpp
|
||||
src/engine/platform/abstract.cpp
|
||||
src/engine/platform/dummy.cpp)
|
||||
|
||||
#imgui/imgui.cpp
|
||||
#imgui/imgui_demo.cpp
|
||||
|
|
|
|||
|
|
@ -8,8 +8,9 @@ void TAAudio::setBufferSizeChangeCallback(void (*callback)(BufferSizeChangeEvent
|
|||
bufferSizeChanged=callback;
|
||||
}
|
||||
|
||||
void TAAudio::setCallback(void (*callback)(float**,float**,int,int,unsigned int)) {
|
||||
void TAAudio::setCallback(void (*callback)(void*,float**,float**,int,int,unsigned int), void* user) {
|
||||
audioProcCallback=callback;
|
||||
audioProcCallbackUser=user;
|
||||
}
|
||||
|
||||
void* TAAudio::getContext() {
|
||||
|
|
|
|||
|
|
@ -33,7 +33,7 @@ void TAAudioJACK::onBufferSize(jack_nframes_t bufsize) {
|
|||
|
||||
void TAAudioJACK::onProcess(jack_nframes_t nframes) {
|
||||
if (audioProcCallback!=NULL) {
|
||||
audioProcCallback(inBufs,outBufs,desc.inChans,desc.outChans,desc.bufsize);
|
||||
audioProcCallback(audioProcCallbackUser,inBufs,outBufs,desc.inChans,desc.outChans,desc.bufsize);
|
||||
}
|
||||
for (int i=0; i<desc.inChans; i++) {
|
||||
iInBufs[i]=(float*)jack_port_get_buffer(ai[i],nframes);
|
||||
|
|
|
|||
|
|
@ -8,7 +8,7 @@ void taSDLProcess(void* inst, unsigned char* buf, int nframes) {
|
|||
|
||||
void TAAudioSDL::onProcess(unsigned char* buf, int nframes) {
|
||||
if (audioProcCallback!=NULL) {
|
||||
audioProcCallback(inBufs,outBufs,desc.inChans,desc.outChans,desc.bufsize);
|
||||
audioProcCallback(audioProcCallbackUser,inBufs,outBufs,desc.inChans,desc.outChans,desc.bufsize);
|
||||
}
|
||||
float* fbuf=(float*)buf;
|
||||
for (size_t j=0; j<desc.bufsize; j++) {
|
||||
|
|
|
|||
|
|
@ -52,14 +52,15 @@ class TAAudio {
|
|||
bool running, initialized;
|
||||
float** inBufs;
|
||||
float** outBufs;
|
||||
void (*audioProcCallback)(float**,float**,int,int,unsigned int);
|
||||
void (*audioProcCallback)(void*,float**,float**,int,int,unsigned int);
|
||||
void* audioProcCallbackUser;
|
||||
void (*sampleRateChanged)(SampleRateChangeEvent);
|
||||
void (*bufferSizeChanged)(BufferSizeChangeEvent);
|
||||
public:
|
||||
void setSampleRateChangeCallback(void (*callback)(SampleRateChangeEvent));
|
||||
void setBufferSizeChangeCallback(void (*callback)(BufferSizeChangeEvent));
|
||||
|
||||
void setCallback(void (*callback)(float**,float**,int,int,unsigned int));
|
||||
void setCallback(void (*callback)(void*,float**,float**,int,int,unsigned int), void* user);
|
||||
|
||||
virtual void* getContext();
|
||||
virtual bool quit();
|
||||
|
|
|
|||
344
src/engine/blip_buf.c
Normal file
344
src/engine/blip_buf.c
Normal file
|
|
@ -0,0 +1,344 @@
|
|||
/* blip_buf $vers. http://www.slack.net/~ant/ */
|
||||
|
||||
#include "blip_buf.h"
|
||||
|
||||
#include <assert.h>
|
||||
#include <limits.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
|
||||
/* Library Copyright (C) 2003-2009 Shay Green. This library 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
|
||||
library 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 */
|
||||
|
||||
#if defined (BLARGG_TEST) && BLARGG_TEST
|
||||
#include "blargg_test.h"
|
||||
#endif
|
||||
|
||||
/* Equivalent to ULONG_MAX >= 0xFFFFFFFF00000000.
|
||||
Avoids constants that don't fit in 32 bits. */
|
||||
#if ULONG_MAX/0xFFFFFFFF > 0xFFFFFFFF
|
||||
typedef unsigned long fixed_t;
|
||||
enum { pre_shift = 32 };
|
||||
|
||||
#elif defined(ULLONG_MAX)
|
||||
typedef unsigned long long fixed_t;
|
||||
enum { pre_shift = 32 };
|
||||
|
||||
#else
|
||||
typedef unsigned fixed_t;
|
||||
enum { pre_shift = 0 };
|
||||
|
||||
#endif
|
||||
|
||||
enum { time_bits = pre_shift + 20 };
|
||||
|
||||
static fixed_t const time_unit = (fixed_t) 1 << time_bits;
|
||||
|
||||
enum { bass_shift = 9 }; /* affects high-pass filter breakpoint frequency */
|
||||
enum { end_frame_extra = 2 }; /* allows deltas slightly after frame length */
|
||||
|
||||
enum { half_width = 8 };
|
||||
enum { buf_extra = half_width*2 + end_frame_extra };
|
||||
enum { phase_bits = 5 };
|
||||
enum { phase_count = 1 << phase_bits };
|
||||
enum { delta_bits = 15 };
|
||||
enum { delta_unit = 1 << delta_bits };
|
||||
enum { frac_bits = time_bits - pre_shift };
|
||||
|
||||
/* We could eliminate avail and encode whole samples in offset, but that would
|
||||
limit the total buffered samples to blip_max_frame. That could only be
|
||||
increased by decreasing time_bits, which would reduce resample ratio accuracy.
|
||||
*/
|
||||
|
||||
/** Sample buffer that resamples to output rate and accumulates samples
|
||||
until they're read out */
|
||||
struct blip_t
|
||||
{
|
||||
fixed_t factor;
|
||||
fixed_t offset;
|
||||
int avail;
|
||||
int size;
|
||||
int integrator;
|
||||
};
|
||||
|
||||
typedef int buf_t;
|
||||
|
||||
/* probably not totally portable */
|
||||
#define SAMPLES( buf ) ((buf_t*) ((buf) + 1))
|
||||
|
||||
/* Arithmetic (sign-preserving) right shift */
|
||||
#define ARITH_SHIFT( n, shift ) \
|
||||
((n) >> (shift))
|
||||
|
||||
enum { max_sample = +32767 };
|
||||
enum { min_sample = -32768 };
|
||||
|
||||
#define CLAMP( n ) \
|
||||
{\
|
||||
if ( (short) n != n )\
|
||||
n = ARITH_SHIFT( n, 16 ) ^ max_sample;\
|
||||
}
|
||||
|
||||
static void check_assumptions( void )
|
||||
{
|
||||
int n;
|
||||
|
||||
#if INT_MAX < 0x7FFFFFFF || UINT_MAX < 0xFFFFFFFF
|
||||
#error "int must be at least 32 bits"
|
||||
#endif
|
||||
|
||||
assert( (-3 >> 1) == -2 ); /* right shift must preserve sign */
|
||||
|
||||
n = max_sample * 2;
|
||||
CLAMP( n );
|
||||
assert( n == max_sample );
|
||||
|
||||
n = min_sample * 2;
|
||||
CLAMP( n );
|
||||
assert( n == min_sample );
|
||||
|
||||
assert( blip_max_ratio <= time_unit );
|
||||
assert( blip_max_frame <= (fixed_t) -1 >> time_bits );
|
||||
}
|
||||
|
||||
blip_t* blip_new( int size )
|
||||
{
|
||||
blip_t* m;
|
||||
assert( size >= 0 );
|
||||
|
||||
m = (blip_t*) malloc( sizeof *m + (size + buf_extra) * sizeof (buf_t) );
|
||||
if ( m )
|
||||
{
|
||||
m->factor = time_unit / blip_max_ratio;
|
||||
m->size = size;
|
||||
blip_clear( m );
|
||||
check_assumptions();
|
||||
}
|
||||
return m;
|
||||
}
|
||||
|
||||
void blip_delete( blip_t* m )
|
||||
{
|
||||
if ( m != NULL )
|
||||
{
|
||||
/* Clear fields in case user tries to use after freeing */
|
||||
memset( m, 0, sizeof *m );
|
||||
free( m );
|
||||
}
|
||||
}
|
||||
|
||||
void blip_set_rates( blip_t* m, double clock_rate, double sample_rate )
|
||||
{
|
||||
double factor = time_unit * sample_rate / clock_rate;
|
||||
m->factor = (fixed_t) factor;
|
||||
|
||||
/* Fails if clock_rate exceeds maximum, relative to sample_rate */
|
||||
assert( 0 <= factor - m->factor && factor - m->factor < 1 );
|
||||
|
||||
/* Avoid requiring math.h. Equivalent to
|
||||
m->factor = (int) ceil( factor ) */
|
||||
if ( m->factor < factor )
|
||||
m->factor++;
|
||||
|
||||
/* At this point, factor is most likely rounded up, but could still
|
||||
have been rounded down in the floating-point calculation. */
|
||||
}
|
||||
|
||||
void blip_clear( blip_t* m )
|
||||
{
|
||||
/* We could set offset to 0, factor/2, or factor-1. 0 is suitable if
|
||||
factor is rounded up. factor-1 is suitable if factor is rounded down.
|
||||
Since we don't know rounding direction, factor/2 accommodates either,
|
||||
with the slight loss of showing an error in half the time. Since for
|
||||
a 64-bit factor this is years, the halving isn't a problem. */
|
||||
|
||||
m->offset = m->factor / 2;
|
||||
m->avail = 0;
|
||||
m->integrator = 0;
|
||||
memset( SAMPLES( m ), 0, (m->size + buf_extra) * sizeof (buf_t) );
|
||||
}
|
||||
|
||||
int blip_clocks_needed( const blip_t* m, int samples )
|
||||
{
|
||||
fixed_t needed;
|
||||
|
||||
/* Fails if buffer can't hold that many more samples */
|
||||
assert( samples >= 0 && m->avail + samples <= m->size );
|
||||
|
||||
needed = (fixed_t) samples * time_unit;
|
||||
if ( needed < m->offset )
|
||||
return 0;
|
||||
|
||||
return (needed - m->offset + m->factor - 1) / m->factor;
|
||||
}
|
||||
|
||||
void blip_end_frame( blip_t* m, unsigned t )
|
||||
{
|
||||
fixed_t off = t * m->factor + m->offset;
|
||||
m->avail += off >> time_bits;
|
||||
m->offset = off & (time_unit - 1);
|
||||
|
||||
/* Fails if buffer size was exceeded */
|
||||
assert( m->avail <= m->size );
|
||||
}
|
||||
|
||||
int blip_samples_avail( const blip_t* m )
|
||||
{
|
||||
return m->avail;
|
||||
}
|
||||
|
||||
static void remove_samples( blip_t* m, int count )
|
||||
{
|
||||
buf_t* buf = SAMPLES( m );
|
||||
int remain = m->avail + buf_extra - count;
|
||||
m->avail -= count;
|
||||
|
||||
memmove( &buf [0], &buf [count], remain * sizeof buf [0] );
|
||||
memset( &buf [remain], 0, count * sizeof buf [0] );
|
||||
}
|
||||
|
||||
int blip_read_samples( blip_t* m, short out [], int count, int stereo )
|
||||
{
|
||||
assert( count >= 0 );
|
||||
|
||||
if ( count > m->avail )
|
||||
count = m->avail;
|
||||
|
||||
if ( count )
|
||||
{
|
||||
int const step = stereo ? 2 : 1;
|
||||
buf_t const* in = SAMPLES( m );
|
||||
buf_t const* end = in + count;
|
||||
int sum = m->integrator;
|
||||
do
|
||||
{
|
||||
/* Eliminate fraction */
|
||||
int s = ARITH_SHIFT( sum, delta_bits );
|
||||
|
||||
sum += *in++;
|
||||
|
||||
CLAMP( s );
|
||||
|
||||
*out = s;
|
||||
out += step;
|
||||
|
||||
/* High-pass filter */
|
||||
sum -= s << (delta_bits - bass_shift);
|
||||
}
|
||||
while ( in != end );
|
||||
m->integrator = sum;
|
||||
|
||||
remove_samples( m, count );
|
||||
}
|
||||
|
||||
return count;
|
||||
}
|
||||
|
||||
/* Things that didn't help performance on x86:
|
||||
__attribute__((aligned(128)))
|
||||
#define short int
|
||||
restrict
|
||||
*/
|
||||
|
||||
/* Sinc_Generator( 0.9, 0.55, 4.5 ) */
|
||||
static short const bl_step [phase_count + 1] [half_width] =
|
||||
{
|
||||
{ 43, -115, 350, -488, 1136, -914, 5861,21022},
|
||||
{ 44, -118, 348, -473, 1076, -799, 5274,21001},
|
||||
{ 45, -121, 344, -454, 1011, -677, 4706,20936},
|
||||
{ 46, -122, 336, -431, 942, -549, 4156,20829},
|
||||
{ 47, -123, 327, -404, 868, -418, 3629,20679},
|
||||
{ 47, -122, 316, -375, 792, -285, 3124,20488},
|
||||
{ 47, -120, 303, -344, 714, -151, 2644,20256},
|
||||
{ 46, -117, 289, -310, 634, -17, 2188,19985},
|
||||
{ 46, -114, 273, -275, 553, 117, 1758,19675},
|
||||
{ 44, -108, 255, -237, 471, 247, 1356,19327},
|
||||
{ 43, -103, 237, -199, 390, 373, 981,18944},
|
||||
{ 42, -98, 218, -160, 310, 495, 633,18527},
|
||||
{ 40, -91, 198, -121, 231, 611, 314,18078},
|
||||
{ 38, -84, 178, -81, 153, 722, 22,17599},
|
||||
{ 36, -76, 157, -43, 80, 824, -241,17092},
|
||||
{ 34, -68, 135, -3, 8, 919, -476,16558},
|
||||
{ 32, -61, 115, 34, -60, 1006, -683,16001},
|
||||
{ 29, -52, 94, 70, -123, 1083, -862,15422},
|
||||
{ 27, -44, 73, 106, -184, 1152,-1015,14824},
|
||||
{ 25, -36, 53, 139, -239, 1211,-1142,14210},
|
||||
{ 22, -27, 34, 170, -290, 1261,-1244,13582},
|
||||
{ 20, -20, 16, 199, -335, 1301,-1322,12942},
|
||||
{ 18, -12, -3, 226, -375, 1331,-1376,12293},
|
||||
{ 15, -4, -19, 250, -410, 1351,-1408,11638},
|
||||
{ 13, 3, -35, 272, -439, 1361,-1419,10979},
|
||||
{ 11, 9, -49, 292, -464, 1362,-1410,10319},
|
||||
{ 9, 16, -63, 309, -483, 1354,-1383, 9660},
|
||||
{ 7, 22, -75, 322, -496, 1337,-1339, 9005},
|
||||
{ 6, 26, -85, 333, -504, 1312,-1280, 8355},
|
||||
{ 4, 31, -94, 341, -507, 1278,-1205, 7713},
|
||||
{ 3, 35, -102, 347, -506, 1238,-1119, 7082},
|
||||
{ 1, 40, -110, 350, -499, 1190,-1021, 6464},
|
||||
{ 0, 43, -115, 350, -488, 1136, -914, 5861}
|
||||
};
|
||||
|
||||
/* Shifting by pre_shift allows calculation using unsigned int rather than
|
||||
possibly-wider fixed_t. On 32-bit platforms, this is likely more efficient.
|
||||
And by having pre_shift 32, a 32-bit platform can easily do the shift by
|
||||
simply ignoring the low half. */
|
||||
|
||||
void blip_add_delta( blip_t* m, unsigned time, int delta )
|
||||
{
|
||||
unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
|
||||
buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
|
||||
|
||||
int const phase_shift = frac_bits - phase_bits;
|
||||
int phase = fixed >> phase_shift & (phase_count - 1);
|
||||
short const* in = bl_step [phase];
|
||||
short const* rev = bl_step [phase_count - phase];
|
||||
|
||||
int interp = fixed >> (phase_shift - delta_bits) & (delta_unit - 1);
|
||||
int delta2 = (delta * interp) >> delta_bits;
|
||||
delta -= delta2;
|
||||
|
||||
/* Fails if buffer size was exceeded */
|
||||
assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
|
||||
|
||||
out [0] += in[0]*delta + in[half_width+0]*delta2;
|
||||
out [1] += in[1]*delta + in[half_width+1]*delta2;
|
||||
out [2] += in[2]*delta + in[half_width+2]*delta2;
|
||||
out [3] += in[3]*delta + in[half_width+3]*delta2;
|
||||
out [4] += in[4]*delta + in[half_width+4]*delta2;
|
||||
out [5] += in[5]*delta + in[half_width+5]*delta2;
|
||||
out [6] += in[6]*delta + in[half_width+6]*delta2;
|
||||
out [7] += in[7]*delta + in[half_width+7]*delta2;
|
||||
|
||||
in = rev;
|
||||
out [ 8] += in[7]*delta + in[7-half_width]*delta2;
|
||||
out [ 9] += in[6]*delta + in[6-half_width]*delta2;
|
||||
out [10] += in[5]*delta + in[5-half_width]*delta2;
|
||||
out [11] += in[4]*delta + in[4-half_width]*delta2;
|
||||
out [12] += in[3]*delta + in[3-half_width]*delta2;
|
||||
out [13] += in[2]*delta + in[2-half_width]*delta2;
|
||||
out [14] += in[1]*delta + in[1-half_width]*delta2;
|
||||
out [15] += in[0]*delta + in[0-half_width]*delta2;
|
||||
}
|
||||
|
||||
void blip_add_delta_fast( blip_t* m, unsigned time, int delta )
|
||||
{
|
||||
unsigned fixed = (unsigned) ((time * m->factor + m->offset) >> pre_shift);
|
||||
buf_t* out = SAMPLES( m ) + m->avail + (fixed >> frac_bits);
|
||||
|
||||
int interp = fixed >> (frac_bits - delta_bits) & (delta_unit - 1);
|
||||
int delta2 = delta * interp;
|
||||
|
||||
/* Fails if buffer size was exceeded */
|
||||
assert( out <= &SAMPLES( m ) [m->size + end_frame_extra] );
|
||||
|
||||
out [7] += delta * delta_unit - delta2;
|
||||
out [8] += delta2;
|
||||
}
|
||||
72
src/engine/blip_buf.h
Normal file
72
src/engine/blip_buf.h
Normal file
|
|
@ -0,0 +1,72 @@
|
|||
/** \file
|
||||
Sample buffer that resamples from input clock rate to output sample rate */
|
||||
|
||||
/* blip_buf $vers */
|
||||
#ifndef BLIP_BUF_H
|
||||
#define BLIP_BUF_H
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
/** First parameter of most functions is blip_t*, or const blip_t* if nothing
|
||||
is changed. */
|
||||
typedef struct blip_t blip_t;
|
||||
|
||||
/** Creates new buffer that can hold at most sample_count samples. Sets rates
|
||||
so that there are blip_max_ratio clocks per sample. Returns pointer to new
|
||||
buffer, or NULL if insufficient memory. */
|
||||
blip_t* blip_new( int sample_count );
|
||||
|
||||
/** Sets approximate input clock rate and output sample rate. For every
|
||||
clock_rate input clocks, approximately sample_rate samples are generated. */
|
||||
void blip_set_rates( blip_t*, double clock_rate, double sample_rate );
|
||||
|
||||
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
|
||||
250
src/engine/blip_buf.txt
Normal file
250
src/engine/blip_buf.txt
Normal file
|
|
@ -0,0 +1,250 @@
|
|||
blip_buf $vers
|
||||
--------------
|
||||
Author : Shay Green <gblargg@gmail.com>
|
||||
Website : http://www.slack.net/~ant/
|
||||
License : GNU Lesser General Public License (LGPL)
|
||||
|
||||
|
||||
Contents
|
||||
--------
|
||||
* Overview
|
||||
* Buffer creation
|
||||
* Waveform generation
|
||||
* Time frames
|
||||
* Complex waveforms
|
||||
* Sample buffering
|
||||
* Thanks
|
||||
|
||||
|
||||
Overview
|
||||
--------
|
||||
This library resamples audio waveforms from input clock rate to output
|
||||
sample rate. Usage follows this general pattern:
|
||||
|
||||
* Create buffer with blip_new().
|
||||
* Set clock rate and sample rate with blip_set_rates().
|
||||
* Waveform generation loop:
|
||||
- Generate several clocks of waveform with blip_add_delta().
|
||||
- End time frame with blip_end_frame().
|
||||
- Read samples from buffer with blip_read_samples().
|
||||
* Free buffer with blip_delete().
|
||||
|
||||
|
||||
Buffer creation
|
||||
---------------
|
||||
Before synthesis, a buffer must be created with blip_new(). Its size is
|
||||
the maximum number of unread samples it can hold. For most uses, this
|
||||
can be 1/10 the sample rate or less, since samples will usually be read
|
||||
out immediately after being generated.
|
||||
|
||||
After the buffer is created, the input clock rate and output sample rate
|
||||
must be set with blip_set_rates(). This determines how many input clocks
|
||||
there are per second, and how many output samples are generated per
|
||||
second.
|
||||
|
||||
If the compiler supports a 64-bit integer type, then the input-output
|
||||
ratio is stored very accurately. If the compiler only supports a 32-bit
|
||||
integer type, then the ratio is stored with only 20 fraction bits, so
|
||||
some ratios cannot be represented exactly (for example, sample
|
||||
rate=48000 and clock rate=48001). The ratio is internally rounded up, so
|
||||
there will never be fewer than 'sample rate' samples per second. Having
|
||||
too many per second is generally better than having too few.
|
||||
|
||||
|
||||
Waveform generation
|
||||
-------------------
|
||||
Waveforms are generated at the input clock rate. Consider a simple
|
||||
square wave with 8 clocks per cycle (4 clocks high, 4 clocks low):
|
||||
|
||||
|<-- 8 clocks ->|
|
||||
+5| ._._._._ ._._._._ ._._._._ ._._
|
||||
| | | | | | | |
|
||||
Amp 0|._._._._ | | | | | |
|
||||
| | | | | | |
|
||||
-5| ._._._._ ._._._._ ._._._._
|
||||
* . . . * . . . * . . . * . . . * . . . * . . . * . . . * .
|
||||
Time 0 4 8 12 16 20 24 28
|
||||
|
||||
The wave changes amplitude at time points 0, 4, 8, 12, 16, etc.
|
||||
|
||||
The following generates the amplitude at every clock of above waveform
|
||||
at the input clock rate:
|
||||
|
||||
int wave [30];
|
||||
|
||||
for ( int i = 4; i < 30; ++i )
|
||||
{
|
||||
if ( i % 8 < 4 )
|
||||
wave [i] = -5;
|
||||
else
|
||||
wave [i] = +5;
|
||||
}
|
||||
|
||||
Without this library, the wave array would then need to be resampled
|
||||
from the input clock rate to the output sample rate. This library does
|
||||
this resampling internally, so it won't be discussed further; waveform
|
||||
generation code can focus entirely on the input clocks.
|
||||
|
||||
Rather than specify the amplitude at every clock, this library merely
|
||||
needs to know the points where the amplitude CHANGES, referred to as a
|
||||
delta. The time of a delta is specified with a clock count. The deltas
|
||||
for this square wave are shown below the time points they occur at:
|
||||
|
||||
+5| ._._._._ ._._._._ ._._._._ ._._
|
||||
| | | | | | | |
|
||||
Amp 0|._._._._ | | | | | |
|
||||
| | | | | | |
|
||||
-5| ._._._._ ._._._._ ._._._._
|
||||
* . . . * . . . * . . . * . . . * . . . * . . . * . . . * .
|
||||
Time 0 4 8 12 16 20 24 28
|
||||
Delta +5 -10 +10 -10 +10 -10 +10
|
||||
|
||||
The following calls generate the above waveform:
|
||||
|
||||
blip_add_delta( blip, 4, +5 );
|
||||
blip_add_delta( blip, 8, -10 );
|
||||
blip_add_delta( blip, 12, +10 );
|
||||
blip_add_delta( blip, 16, -10 );
|
||||
blip_add_delta( blip, 20, +10 );
|
||||
blip_add_delta( blip, 24, -10 );
|
||||
blip_add_delta( blip, 28, +10 );
|
||||
|
||||
In the examples above, the amplitudes are small for clarity. The 16-bit
|
||||
sample range is -32768 to +32767, so actual waveform amplitudes would
|
||||
need to be in the thousands to be audible (for example, -5000 to +5000).
|
||||
|
||||
This library allows waveform generation code to pay NO attention to the
|
||||
output sample rate. It can focus ENTIRELY on the essence of the
|
||||
waveform: the points where its amplitude changes. Since these points can
|
||||
be efficiently generated in a loop, synthesis is efficient. Sound chip
|
||||
emulation code can be structured to allow full accuracy down to a single
|
||||
clock, with the emulated CPU being able to simply tell the sound chip to
|
||||
"emulate from wherever you left off, up to clock time T within the
|
||||
current time frame".
|
||||
|
||||
|
||||
Time frames
|
||||
-----------
|
||||
Since time keeps increasing, if left unchecked, at some point it would
|
||||
overflow the range of an integer. This library's solution to the problem
|
||||
is to break waveform generation into time frames of moderate length.
|
||||
Clock counts within a time frame are thus relative to the beginning of
|
||||
the frame, where 0 is the beginning of the frame. When a time frame of
|
||||
length T is ended, what was at time T in the old time frame is now at
|
||||
time 0 in the new time frame. Breaking the above waveform into time
|
||||
frames of 10 clocks each looks like this:
|
||||
|
||||
+5| ._._._._ ._._._._ ._._._._ ._._
|
||||
| | | | | | | |
|
||||
Amp 0|._._._._ | | | | | |
|
||||
| | | | | | |
|
||||
-5| ._._._._ ._._._._ ._._._._
|
||||
* . . . * . . . * . . . * . . . * . . . * . . . * . . . * .
|
||||
Time |0 4 8 | 2 6 |0 4 8 |
|
||||
| first time frame | second time frame | third time frame |
|
||||
|<--- 10 clocks --->|<--- 10 clocks --->|<--- 10 clocks --->|
|
||||
|
||||
The following calls generate the above waveform. After they execute, the
|
||||
first 30 clocks of the waveform will have been resampled and be
|
||||
available as output samples for reading with blip_read_samples().
|
||||
|
||||
blip_add_delta( blip, 4, +5 );
|
||||
blip_add_delta( blip, 8, -10 );
|
||||
blip_end_frame( blip, 10 );
|
||||
|
||||
blip_add_delta( blip, 2, +10 );
|
||||
blip_add_delta( blip, 6, -10 );
|
||||
blip_end_frame( blip, 10 );
|
||||
|
||||
blip_add_delta( blip, 0, +10 );
|
||||
blip_add_delta( blip, 4, -10 );
|
||||
blip_add_delta( blip, 8, +10 );
|
||||
blip_end_frame( blip, 10 );
|
||||
...
|
||||
|
||||
Time frames can be a convenient length, and the length can vary from one
|
||||
frame to the next. Once a time frame is ended, the resulting output
|
||||
samples become available for reading immediately, and no more deltas can
|
||||
be added to it.
|
||||
|
||||
There is a limit of about 4000 output samples per time frame. The number
|
||||
of clocks depends on the clock rate. At common sample rates, this allows
|
||||
time frames of at least 1/15 second, plenty for most uses. This limit
|
||||
allows increased resampling ratio accuracy.
|
||||
|
||||
In an emulator, it is usually convenient to have audio time frames
|
||||
correspond to video frames, where the CPU's clock counter is reset at
|
||||
the beginning of each video frame and thus can be used directly as the
|
||||
relative clock counts for audio time frames.
|
||||
|
||||
|
||||
Complex waveforms
|
||||
-----------------
|
||||
Any sort of waveform can be generated, not just a square wave. For
|
||||
example, a saw-like wave:
|
||||
|
||||
+5| ._._._._ ._._._._ ._._
|
||||
| | | | | |
|
||||
Amp 0|._._._._ | ._._._._ | ._._._._
|
||||
| | | | |
|
||||
-5| ._._._._ ._._._._
|
||||
* . . . * . . . * . . . * . . . * . . . * . . . * . . . * .
|
||||
Time 0 4 8 12 16 20 24 28
|
||||
Delta +5 -10 +5 +5 -10 +5 +5
|
||||
|
||||
Code to generate above waveform:
|
||||
|
||||
blip_add_delta( blip, 4, +5 );
|
||||
blip_add_delta( blip, 8, -10 );
|
||||
blip_add_delta( blip, 12, +5 );
|
||||
blip_add_delta( blip, 16, +5 );
|
||||
blip_add_delta( blip, 20, +10 );
|
||||
blip_add_delta( blip, 24, +5 );
|
||||
blip_add_delta( blip, 28, +5 );
|
||||
|
||||
Similarly, multiple waveforms can be added within a time frame without
|
||||
problem. It doesn't matter what order they're added, because all the
|
||||
library needs are the deltas. The synthesis code doesn't need to know
|
||||
all the waveforms at once either; it can calculate and add the deltas
|
||||
for each waveform individually. Deltas don't need to be added in
|
||||
chronological order either.
|
||||
|
||||
|
||||
Sample buffering
|
||||
----------------
|
||||
Sample buffering is very flexible. Once a time frame is ended, the
|
||||
resampled waveforms become output samples that are immediately made
|
||||
available for reading with blip_read_samples(). They don't have to be
|
||||
read immediately; they can be allowed to accumulate in the buffer, with
|
||||
each time frame appending more samples to the buffer. When reading, some
|
||||
or all of the samples in can be read out, with the remaining unread
|
||||
samples staying in the buffer for later. Usually a program will
|
||||
immediately read all available samples after ending a time frame and
|
||||
play them immediately. In some systems, a program needs samples in
|
||||
fixed-length blocks; in that case, it would keep generating time frames
|
||||
until some number of samples are available, then read only that many,
|
||||
even if slightly more were available in the buffer.
|
||||
|
||||
In some systems, one wants to run waveform generation for exactly the
|
||||
number of clocks necessary to generate some desired number of output
|
||||
samples, and no more. In that case, use blip_clocks_needed( blip, N ) to
|
||||
find out how many clocks are needed to generate N additional samples.
|
||||
Ending a time frame with this value will result in exactly N more
|
||||
samples becoming available for reading.
|
||||
|
||||
|
||||
Thanks
|
||||
------
|
||||
Thanks to Jsr (FamiTracker author), the Mednafen team (multi-system
|
||||
emulator), ShizZie (Nhes GMB author), Marcel van Tongeren, Luke Molnar
|
||||
(UberNES author), Fredrick Meunier (Fuse contributor) for using and
|
||||
giving feedback for another similar library. Thanks to Disch for his
|
||||
interest and discussions about the synthesis algorithm itself, and for
|
||||
writing his own implementation of it (Schpune) rather than just using
|
||||
mine. Thanks to Xodnizel for Festalon, whose sound quality got me
|
||||
interested in video game sound emulation in the first place, and where I
|
||||
first came up with the algorithm while optimizing its brute-force
|
||||
filter.
|
||||
|
||||
--
|
||||
Shay Green <gblargg@gmail.com>
|
||||
|
|
@ -1,3 +1,6 @@
|
|||
#ifndef _DISPATCH_H
|
||||
#define _DISPATCH_H
|
||||
|
||||
enum DivDispatchCmds {
|
||||
DIV_CMD_NOTE_ON=0,
|
||||
DIV_CMD_NOTE_OFF,
|
||||
|
|
@ -17,15 +20,27 @@ struct DivDelayedCommand {
|
|||
DivCommand cmd;
|
||||
};
|
||||
|
||||
class DivEngine;
|
||||
|
||||
class DivDispatch {
|
||||
protected:
|
||||
DivEngine* parent;
|
||||
public:
|
||||
virtual void acquire(float& l, float& r);
|
||||
/**
|
||||
* the rate the samples are provided.
|
||||
* the engine shall resample to the output rate.
|
||||
*/
|
||||
int rate;
|
||||
virtual void acquire(short& l, short& r);
|
||||
virtual int dispatch(DivCommand c);
|
||||
|
||||
/**
|
||||
* initialize this DivDispatch.
|
||||
* @param parent the parent DivEngine.
|
||||
* @param channels the number of channels to acquire.
|
||||
* @param sugRate the suggested rate. this may change, so don't rely on it.
|
||||
* @return the number of channels allocated.
|
||||
*/
|
||||
virtual int init(int channels);
|
||||
virtual int init(DivEngine* parent, int channels, int sugRate);
|
||||
};
|
||||
#endif
|
||||
|
|
@ -2,16 +2,11 @@
|
|||
#include "safeReader.h"
|
||||
#include "../ta-log.h"
|
||||
#include "../audio/sdl.h"
|
||||
#include "platform/dummy.h"
|
||||
#include <zlib.h>
|
||||
|
||||
void process(float** in, float** out, int inChans, int outChans, unsigned int size) {
|
||||
static int count;
|
||||
for (unsigned int i=0; i<size; i++) {
|
||||
count++;
|
||||
for (int j=0; j<outChans; j++) {
|
||||
out[j][i]=((count%160)>40)?0.5:0.0;
|
||||
}
|
||||
}
|
||||
void process(void* u, float** in, float** out, int inChans, int outChans, unsigned int size) {
|
||||
((DivEngine*)u)->nextBuf(in,out,inChans,outChans,size);
|
||||
}
|
||||
|
||||
#define DIV_READ_SIZE 131072
|
||||
|
|
@ -596,6 +591,7 @@ bool DivEngine::load(void* f, size_t slen) {
|
|||
}
|
||||
|
||||
song=ds;
|
||||
chans=getChannelCount(song.system);
|
||||
} catch (EndOfFileException e) {
|
||||
logE("premature end of file!\n");
|
||||
return false;
|
||||
|
|
@ -617,7 +613,7 @@ bool DivEngine::init() {
|
|||
want.outFormat=TA_AUDIO_FORMAT_F32;
|
||||
want.name="DivAudio";
|
||||
|
||||
output->setCallback(process);
|
||||
output->setCallback(process,this);
|
||||
|
||||
logI("initializing audio.\n");
|
||||
if (!output->init(want,got)) {
|
||||
|
|
@ -625,8 +621,29 @@ bool DivEngine::init() {
|
|||
return false;
|
||||
}
|
||||
|
||||
bb[0]=blip_new(32768);
|
||||
if (bb[0]==NULL) {
|
||||
logE("not enough memory!\n");
|
||||
return false;
|
||||
}
|
||||
|
||||
bb[1]=blip_new(32768);
|
||||
if (bb[1]==NULL) {
|
||||
logE("not enough memory!\n");
|
||||
return false;
|
||||
}
|
||||
|
||||
bbOut[0]=new short[got.bufsize];
|
||||
bbOut[1]=new short[got.bufsize];
|
||||
|
||||
dispatch=new DivPlatformDummy;
|
||||
dispatch->init(this,getChannelCount(song.system),got.rate);
|
||||
|
||||
blip_set_rates(bb[0],dispatch->rate,got.rate);
|
||||
blip_set_rates(bb[1],dispatch->rate,got.rate);
|
||||
|
||||
if (!output->setRun(true)) {
|
||||
printf("error while activating!\n");
|
||||
logE("error while activating!\n");
|
||||
return false;
|
||||
}
|
||||
return true;
|
||||
|
|
|
|||
|
|
@ -1,6 +1,9 @@
|
|||
#ifndef _ENGINE_H
|
||||
#define _ENGINE_H
|
||||
#include "song.h"
|
||||
#include "dispatch.h"
|
||||
#include "../audio/taAudio.h"
|
||||
#include "blip_buf.h"
|
||||
|
||||
struct DivChannelState {
|
||||
std::vector<DivDelayedCommand> delayed;
|
||||
|
|
@ -18,10 +21,19 @@ class DivEngine {
|
|||
int chans;
|
||||
bool playing;
|
||||
bool speedAB;
|
||||
int ticks, curRow, curOrder;
|
||||
int ticks, cycles, curRow, curOrder;
|
||||
std::vector<DivChannelState> chan;
|
||||
|
||||
blip_buffer_t* bb[2];
|
||||
short temp[2], prevSample[2];
|
||||
short* bbOut[2];
|
||||
|
||||
void nextOrder();
|
||||
void nextRow();
|
||||
void nextTick();
|
||||
|
||||
public:
|
||||
void nextBuf(float** in, float** out, int inChans, int outChans, unsigned int size);
|
||||
// load a .dmf.
|
||||
bool load(void* f, size_t length);
|
||||
// save as .dmf.
|
||||
|
|
@ -32,4 +44,16 @@ class DivEngine {
|
|||
|
||||
// initialize the engine.
|
||||
bool init();
|
||||
|
||||
DivEngine():
|
||||
chans(0),
|
||||
playing(false),
|
||||
speedAB(false),
|
||||
ticks(0),
|
||||
cycles(0),
|
||||
curRow(-1),
|
||||
curOrder(0),
|
||||
temp{0,0},
|
||||
prevSample{0,0} {}
|
||||
};
|
||||
#endif
|
||||
14
src/engine/platform/abstract.cpp
Normal file
14
src/engine/platform/abstract.cpp
Normal file
|
|
@ -0,0 +1,14 @@
|
|||
#include "../dispatch.h"
|
||||
|
||||
void DivDispatch::acquire(short& l, short& r) {
|
||||
l=0;
|
||||
r=0;
|
||||
}
|
||||
|
||||
int DivDispatch::dispatch(DivCommand c) {
|
||||
return 1;
|
||||
}
|
||||
|
||||
int DivDispatch::init(DivEngine* p, int channels, int sugRate) {
|
||||
return 0;
|
||||
}
|
||||
|
|
@ -0,0 +1,16 @@
|
|||
#include "dummy.h"
|
||||
|
||||
void DivPlatformDummy::acquire(short& l, short& r) {
|
||||
l=0;
|
||||
r=0;
|
||||
}
|
||||
|
||||
int DivPlatformDummy::dispatch(DivCommand c) {
|
||||
return 1;
|
||||
}
|
||||
|
||||
int DivPlatformDummy::init(DivEngine* p, int channels, int sugRate) {
|
||||
parent=p;
|
||||
rate=sugRate;
|
||||
return channels;
|
||||
}
|
||||
|
|
@ -4,7 +4,7 @@
|
|||
// used when a DivDispatch for a system is not found.
|
||||
class DivPlatformDummy: public DivDispatch {
|
||||
public:
|
||||
void acquire(float& l, float& r);
|
||||
void acquire(short& l, short& r);
|
||||
int dispatch(DivCommand c);
|
||||
int init(int channels);
|
||||
int init(DivEngine* parent, int channels, int sugRate);
|
||||
};
|
||||
118
src/engine/playback.cpp
Normal file
118
src/engine/playback.cpp
Normal file
|
|
@ -0,0 +1,118 @@
|
|||
#include "engine.h"
|
||||
|
||||
void DivEngine::nextOrder() {
|
||||
curRow=0;
|
||||
if (++curOrder>=song.ordersLen) {
|
||||
curOrder=0;
|
||||
}
|
||||
}
|
||||
|
||||
const char* notes[12]={
|
||||
"C-", "C#", "D-", "D#", "E-", "F-", "F#", "G-", "G#", "A-", "A#", "B-"
|
||||
};
|
||||
|
||||
const char* formatNote(unsigned char note, unsigned char octave) {
|
||||
static char ret[4];
|
||||
if (note==100) {
|
||||
return "OFF";
|
||||
} else if (octave==0) {
|
||||
return "---";
|
||||
}
|
||||
snprintf(ret,4,"%s%d",notes[note%12],octave+note/12);
|
||||
return ret;
|
||||
}
|
||||
|
||||
void DivEngine::nextRow() {
|
||||
static char pb[4096];
|
||||
static char pb1[4096];
|
||||
static char pb2[4096];
|
||||
static char pb3[4096];
|
||||
if (++curRow>=song.patLen) {
|
||||
nextOrder();
|
||||
}
|
||||
strcpy(pb1,"");
|
||||
strcpy(pb3,"");
|
||||
for (int i=0; i<chans; i++) {
|
||||
snprintf(pb,4095," %.2x",song.orders.ord[i][curOrder]);
|
||||
strcat(pb1,pb);
|
||||
|
||||
DivPattern* pat=song.pat[i]->data[curOrder];
|
||||
snprintf(pb2,4095,"\x1b[37m %s",
|
||||
formatNote(pat->data[curRow][0],pat->data[curRow][1]));
|
||||
strcat(pb3,pb2);
|
||||
if (pat->data[curRow][3]==255) {
|
||||
strcat(pb3,"\x1b[m--");
|
||||
} else {
|
||||
snprintf(pb2,4095,"\x1b[1;32m%.2x",pat->data[curRow][3]);
|
||||
strcat(pb3,pb2);
|
||||
}
|
||||
if (pat->data[curRow][2]==255) {
|
||||
strcat(pb3,"\x1b[m--");
|
||||
} else {
|
||||
snprintf(pb2,4095,"\x1b[0;36m%.2x",pat->data[curRow][2]);
|
||||
strcat(pb3,pb2);
|
||||
}
|
||||
for (int j=0; j<song.pat[i]->effectRows; j++) {
|
||||
if (pat->data[curRow][4+(j<<1)]==255) {
|
||||
strcat(pb3,"\x1b[m--");
|
||||
} else {
|
||||
snprintf(pb2,4095,"\x1b[1;31m%.2x",pat->data[curRow][4+(j<<1)]);
|
||||
strcat(pb3,pb2);
|
||||
}
|
||||
if (pat->data[curRow][5+(j<<1)]==255) {
|
||||
strcat(pb3,"\x1b[m--");
|
||||
} else {
|
||||
snprintf(pb2,4095,"\x1b[1;37m%.2x",pat->data[curRow][5+(j<<1)]);
|
||||
strcat(pb3,pb2);
|
||||
}
|
||||
}
|
||||
}
|
||||
printf("| %.2x:%s | \x1b[1;33m%3d%s\x1b[m\n",curOrder,pb1,curRow,pb3);
|
||||
}
|
||||
|
||||
void DivEngine::nextTick() {
|
||||
if (song.customTempo) {
|
||||
cycles=dispatch->rate/song.hz;
|
||||
} else {
|
||||
if (song.pal) {
|
||||
cycles=dispatch->rate/60;
|
||||
} else {
|
||||
cycles=dispatch->rate/50;
|
||||
}
|
||||
}
|
||||
if (--ticks<=0) {
|
||||
if (speedAB) {
|
||||
ticks=song.speed2*(song.timeBase+1);
|
||||
} else {
|
||||
ticks=song.speed1*(song.timeBase+1);
|
||||
}
|
||||
speedAB=!speedAB;
|
||||
nextRow();
|
||||
}
|
||||
}
|
||||
|
||||
void DivEngine::nextBuf(float** in, float** out, int inChans, int outChans, unsigned int size) {
|
||||
size_t runtotal=blip_clocks_needed(bb[0],size);
|
||||
for (size_t i=0; i<runtotal; i++) {
|
||||
if (--cycles<=0) {
|
||||
nextTick();
|
||||
}
|
||||
dispatch->acquire(temp[0],temp[1]);
|
||||
|
||||
blip_add_delta(bb[0],i,temp[0]-prevSample[0]);
|
||||
blip_add_delta(bb[1],i,temp[1]-prevSample[1]);
|
||||
prevSample[0]=temp[0];
|
||||
prevSample[1]=temp[1];
|
||||
}
|
||||
|
||||
blip_end_frame(bb[0],runtotal);
|
||||
blip_end_frame(bb[1],runtotal);
|
||||
|
||||
blip_read_samples(bb[0],bbOut[0],size,0);
|
||||
blip_read_samples(bb[1],bbOut[1],size,0);
|
||||
|
||||
for (size_t i=0; i<size; i++) {
|
||||
out[0][i]=(float)bbOut[0][i]/32768.0;
|
||||
out[1][i]=(float)bbOut[1][i]/32768.0;
|
||||
}
|
||||
}
|
||||
Loading…
Reference in a new issue