Update DUMB to revision 9ac6cf69758fe0db6d6e654f298cd36efdb73366

- Replaced old aliased resampling mode with a 65536x oversampling sinc resampler


SVN r4076 (trunk)
This commit is contained in:
Randy Heit 2013-02-07 23:21:36 +00:00
parent 13ffa8a1b8
commit 401ee8fafb
10 changed files with 682 additions and 85 deletions

View file

@ -36,6 +36,7 @@ add_library( dumb
src/core/rendsig.c
src/core/unload.c
src/helpers/barray.c
src/helpers/blip_buf.c
src/helpers/clickrem.c
src/helpers/memfile.c
src/helpers/resample.c

View file

@ -662,6 +662,8 @@ typedef struct DUMB_VOLUME_RAMP_INFO DUMB_VOLUME_RAMP_INFO;
typedef void (*DUMB_RESAMPLE_PICKUP)(DUMB_RESAMPLER *resampler, void *data);
#include "internal/blip_buf.h"
struct DUMB_RESAMPLER
{
void *src;
@ -679,6 +681,9 @@ struct DUMB_RESAMPLER
signed char x8[3*2];
} x;
int overshot;
int last_clock;
int last_amp[2];
blip_t* blip_buffer[2];
};
struct DUMB_VOLUME_RAMP_INFO

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@ -0,0 +1,77 @@
/** \file
Sample buffer that resamples from input clock rate to output sample rate */
/* blip_buf 1.1.0 */
#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 );
blip_t* blip_dup( blip_t* );
/** 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*, int out [], int count );
/** Reads the current integrator and returns it */
int blip_peek_sample( blip_t* );
/** Frees buffer. No effect if NULL is passed. */
void blip_delete( blip_t* );
/* Deprecated */
typedef blip_t blip_buffer_t;
#ifdef __cplusplus
}
#endif
#endif

354
dumb/src/helpers/blip_buf.c Normal file
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@ -0,0 +1,354 @@
/* blip_buf 1.1.0. http://www.slack.net/~ant/ */
#include "internal/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;
}
blip_t* blip_dup( blip_t* m )
{
size_t size = sizeof *m + (m->size + buf_extra) * sizeof(buf_t);
blip_t* r = (blip_t*) malloc( size );
if ( r ) memcpy( r, m, size );
return r;
}
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 (int)((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 += (int)(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, int out [], int count )
{
assert( count >= 0 );
if ( count > m->avail )
count = m->avail;
if ( count )
{
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 - 8 );
sum += *in++;
*out = s;
out++;
/* High-pass filter */
sum -= s >> (8 - (delta_bits - bass_shift)); //<< (delta_bits - bass_shift - 8);
}
while ( in != end );
m->integrator = sum;
remove_samples( m, count );
}
return count;
}
int blip_peek_sample( blip_t* m )
{
return ARITH_SHIFT( m->integrator, delta_bits - 8 );
}
/* 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;
}

View file

@ -95,7 +95,8 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
#define SET_VOLUME_VARIABLES SET_MONO_DEST_VOLUME_VARIABLES
#define RETURN_VOLUME_VARIABLES RETURN_MONO_DEST_VOLUME_VARIABLES
#define VOLUMES_ARE_ZERO MONO_DEST_VOLUMES_ARE_ZERO
#define MIX_ALIAS(op, upd, offset) MONO_DEST_MIX_ALIAS(op, upd, offset)
#define MIX_ALIAS(count) MONO_DEST_MIX_ALIAS(count)
#define PEEK_ALIAS MONO_DEST_PEEK_ALIAS
#define MIX_LINEAR(op, upd, o0, o1) MONO_DEST_MIX_LINEAR(op, upd, o0, o1)
#define MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3) MONO_DEST_MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3)
#define MIX_ZEROS(op) *dst++ op 0
@ -137,7 +138,8 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
if ( volume_right ) volume_right->volume = (float)rvolr / 16777216.0f; \
}
#define VOLUMES_ARE_ZERO (lvol == 0 && lvolt == 0 && rvol == 0 && rvolt == 0)
#define MIX_ALIAS(op, upd, offset) STEREO_DEST_MIX_ALIAS(op, upd, offset)
#define MIX_ALIAS(count) STEREO_DEST_MIX_ALIAS(count)
#define PEEK_ALIAS STEREO_DEST_PEEK_ALIAS
#define MIX_LINEAR(op, upd, o0, o1) STEREO_DEST_MIX_LINEAR(op, upd, o0, o1)
#define MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3) STEREO_DEST_MIX_CUBIC(op, upd, x0, x3, o0, o1, o2, o3)
#define MIX_ZEROS(op) { *dst++ op 0; *dst++ op 0; }
@ -157,6 +159,9 @@ static int process_pickup(DUMB_RESAMPLER *resampler)
#undef MONO_DEST_VOLUME_ZEROS
#undef MONO_DEST_VOLUME_VARIABLES
#undef MONO_DEST_VOLUME_PARAMETERS
#undef STEREO_DEST_PEEK_ALIAS
#undef MONO_DEST_PEEK_ALIAS
#undef POKE_ALIAS
#undef COPYSRC2
#undef COPYSRC
#undef DIVIDE_BY_SRC_CHANNELS

View file

@ -46,12 +46,13 @@
int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VOLUME_PARAMETERS, double delta)
{
int dt;
int dt, inv_dt;
int VOLUME_VARIABLES;
long done;
long todo;
LONG_LONG todo64;
int quality;
int blip_samples[256*SRC_CHANNELS];
if (!resampler || resampler->dir == 0) return 0;
ASSERT(resampler->dir == -1 || resampler->dir == 1);
@ -59,6 +60,7 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
done = 0;
dt = (int)(delta * 65536.0 + 0.5);
if (dt == 0 || dt == 0x80000000) return 0;
inv_dt = (int)(1.0 / delta * 65536.0 + 0.5);
SET_VOLUME_VARIABLES;
if (VOLUMES_ARE_ZERO) dst = NULL;
@ -104,29 +106,34 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
subpos = (long)new_subpos & 65535;
} else if (quality <= DUMB_RQ_ALIASING) {
/* Aliasing, backwards */
SRCTYPE xbuf[2*SRC_CHANNELS];
int todo_clocks = todo << 16, todo_clocks_set = todo_clocks;
SRCTYPE xbuf[2*SRC_CHANNELS];
SRCTYPE *x = &xbuf[0];
SRCTYPE *xstart;
COPYSRC(xbuf, 0, resampler->X, 1);
COPYSRC(xbuf, 1, resampler->X, 2);
while (todo && x < &xbuf[2*SRC_CHANNELS]) {
if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
while (resampler->last_clock < todo_clocks_set && x < &xbuf[2*SRC_CHANNELS]) {
// TODO: check what happens when multiple tempo slides occur per row
HEAVYASSERT(pos >= resampler->start);
MIX_ALIAS(+=, 1, 0);
subpos += dt;
pos += subpos >> 16;
x -= (subpos >> 16) * SRC_CHANNELS;
subpos &= 65535;
todo--;
POKE_ALIAS(0);
pos--;
x += SRC_CHANNELS;
}
x = &src[pos*SRC_CHANNELS];
while ( todo_clocks ) {
todo_clocks_set = todo_clocks;
if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
todo_clocks -= todo_clocks_set;
todo = ( todo_clocks_set - resampler->last_clock + inv_dt - 1 ) / inv_dt;
if ( todo < 0 ) todo = 0;
LOOP4(todo,
POKE_ALIAS(2);
pos--;
x -= SRC_CHANNELS;
);
todo = todo_clocks_set >> 16;
MIX_ALIAS( todo );
}
x = xstart = &src[pos*SRC_CHANNELS];
LOOP4(todo,
MIX_ALIAS(+=, 1, 2);
subpos += dt;
x += (subpos >> 16) * SRC_CHANNELS;
subpos &= 65535;
);
pos += DIVIDE_BY_SRC_CHANNELS((long)(x - xstart));
} else if (quality <= DUMB_RQ_LINEAR) {
/* Linear interpolation, backwards */
SRCTYPE xbuf[3*SRC_CHANNELS];
@ -205,28 +212,33 @@ int32 dumb_resample(DUMB_RESAMPLER *resampler, sample_t *dst, int32 dst_size, VO
subpos = (long)new_subpos & 65535;
} else if (quality <= DUMB_RQ_ALIASING) {
/* Aliasing, forwards */
int todo_clocks = todo << 16, todo_clocks_set = todo_clocks;
SRCTYPE xbuf[2*SRC_CHANNELS];
SRCTYPE *x = &xbuf[0];
SRCTYPE *xstart;
COPYSRC(xbuf, 0, resampler->X, 1);
COPYSRC(xbuf, 1, resampler->X, 2);
while (todo && x < &xbuf[2*SRC_CHANNELS]) {
if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
while (resampler->last_clock < todo_clocks_set && x < &xbuf[2*SRC_CHANNELS]) {
HEAVYASSERT(pos < resampler->end);
MIX_ALIAS(+=, 1, 0);
subpos += dt;
pos += subpos >> 16;
x += (subpos >> 16) * SRC_CHANNELS;
subpos &= 65535;
todo--;
POKE_ALIAS(0);
pos++;
x += SRC_CHANNELS;
}
x = &src[pos*SRC_CHANNELS];
while ( todo_clocks ) {
todo_clocks_set = todo_clocks;
if ( todo_clocks_set > 256 * 65536 ) todo_clocks_set = 256 * 65536;
todo_clocks -= todo_clocks_set;
todo = ( todo_clocks_set - resampler->last_clock + inv_dt - 1 ) / inv_dt;
if ( todo < 0 ) todo = 0;
LOOP4(todo,
POKE_ALIAS(-2);
pos++;
x += SRC_CHANNELS;
);
todo = todo_clocks_set >> 16;
MIX_ALIAS( todo );
}
x = xstart = &src[pos*SRC_CHANNELS];
LOOP4(todo,
MIX_ALIAS(+=, 1, -2);
subpos += dt;
x += (subpos >> 16) * SRC_CHANNELS;
subpos &= 65535;
);
pos += DIVIDE_BY_SRC_CHANNELS((long)(x - xstart));
} else if (quality <= DUMB_RQ_LINEAR) {
/* Linear interpolation, forwards */
SRCTYPE xbuf[3*SRC_CHANNELS];
@ -339,7 +351,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
HEAVYASSERT(pos >= resampler->start);
if (dumb_resampling_quality <= DUMB_RQ_ALIASING) {
/* Aliasing, backwards */
MIX_ALIAS(=, 0, 1);
PEEK_ALIAS;
} else if (quality <= DUMB_RQ_LINEAR) {
/* Linear interpolation, backwards */
MIX_LINEAR(=, 0, 2, 1);
@ -351,7 +363,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
HEAVYASSERT(pos < resampler->end);
if (dumb_resampling_quality <= DUMB_RQ_ALIASING) {
/* Aliasing */
MIX_ALIAS(=, 0, 1);
PEEK_ALIAS;
} else if (dumb_resampling_quality <= DUMB_RQ_LINEAR) {
/* Linear interpolation, forwards */
MIX_LINEAR(=, 0, 1, 2);
@ -368,6 +380,7 @@ void dumb_resample_get_current_sample(DUMB_RESAMPLER *resampler, VOLUME_PARAMETE
#undef MIX_CUBIC
#undef MIX_LINEAR
#undef MIX_ALIAS
#undef PEEK_ALIAS
#undef VOLUMES_ARE_ZERO
#undef SET_VOLUME_VARIABLES
#undef RETURN_VOLUME_VARIABLES

View file

@ -189,7 +189,7 @@ static void init_cubic(void)
#define SRCTYPE sample_t
#define SRCBITS 24
#define ALIAS(x, vol) MULSC(x, vol)
#define ALIAS(x) (x >> 8)
#define LINEAR(x0, x1) (x0 + MULSC(x1 - x0, subpos))
/*
#define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \
@ -225,7 +225,7 @@ static void init_cubic(void)
#define SUFFIX _16
#define SRCTYPE short
#define SRCBITS 16
#define ALIAS(x, vol) (x * vol >> 8)
#define ALIAS(x) (x)
#define LINEAR(x0, x1) ((x0 << 8) + MULSC16(x1 - x0, subpos))
/*
#define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \
@ -247,7 +247,7 @@ static void init_cubic(void)
#define SUFFIX _8
#define SRCTYPE signed char
#define SRCBITS 8
#define ALIAS(x, vol) (x * vol)
#define ALIAS(x) (x << 8)
#define LINEAR(x0, x1) ((x0 << 16) + (x1 - x0) * subpos)
/*
#define SET_CUBIC_COEFFICIENTS(x0, x1, x2, x3) { \

View file

@ -69,6 +69,11 @@ void dumb_reset_resampler(DUMB_RESAMPLER *resampler, SRCTYPE *src, int src_chann
}
for (i = 0; i < src_channels*3; i++) resampler->X[i] = 0;
resampler->overshot = -1;
resampler->last_clock = 0;
resampler->last_amp[0] = 0;
resampler->last_amp[1] = 0;
blip_clear(resampler->blip_buffer[0]);
blip_clear(resampler->blip_buffer[1]);
}
@ -77,6 +82,21 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
{
DUMB_RESAMPLER *resampler = malloc(sizeof(*resampler));
if (!resampler) return NULL;
resampler->blip_buffer[0] = blip_new( 256 );
if (!resampler->blip_buffer[0])
{
free(resampler);
return NULL;
}
resampler->blip_buffer[1] = blip_new( 256 );
if (!resampler->blip_buffer[1])
{
free(resampler->blip_buffer[0]);
free(resampler);
return NULL;
}
blip_set_rates(resampler->blip_buffer[0], 65536, 1);
blip_set_rates(resampler->blip_buffer[1], 65536, 1);
dumb_reset_resampler(resampler, src, src_channels, pos, start, end, quality);
return resampler;
}
@ -123,16 +143,41 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
}
#define RETURN_MONO_DEST_VOLUME_VARIABLES if ( volume ) volume->volume = (float)volr / 16777216.0f
#define MONO_DEST_VOLUMES_ARE_ZERO (vol == 0 && volt == 0)
#define MONO_DEST_MIX_ALIAS(op, upd, offset) { \
*dst++ op ALIAS(x[offset], vol); \
if ( upd ) UPDATE_VOLUME( volume, vol ); \
#define POKE_ALIAS(offset) { \
int delta = ALIAS(x[offset]) - resampler->last_amp[0]; \
resampler->last_amp[0] += delta; \
if ( delta ) blip_add_delta( resampler->blip_buffer[0], resampler->last_clock, delta ); \
resampler->last_clock += inv_dt; \
}
#define STEREO_DEST_MIX_ALIAS(op, upd, offset) { \
int xm = x[offset]; \
*dst++ op ALIAS(xm, lvol); \
*dst++ op ALIAS(xm, rvol); \
if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
#define MONO_DEST_PEEK_ALIAS *dst = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), vol )
#define MONO_DEST_MIX_ALIAS(count) { \
int n = 0; \
resampler->last_clock -= count * 65536; \
blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
LOOP4( count, \
*dst++ += MULSC( blip_samples[n], vol ); \
n++; \
UPDATE_VOLUME( volume, vol ); \
); \
}
#define STEREO_DEST_PEEK_ALIAS { \
int sample = blip_peek_sample( resampler->blip_buffer[0] ); \
*dst++ = MULSC( sample, lvol ); \
*dst++ = MULSC( sample, rvol ); \
}
#define STEREO_DEST_MIX_ALIAS(count) { \
int sample, n = 0; \
resampler->last_clock -= count * 65536; \
blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
LOOP4( count, \
sample = blip_samples[n++]; \
*dst++ += MULSC( sample, lvol ); \
*dst++ += MULSC( sample, rvol ); \
UPDATE_VOLUME( volume_left, lvol ); \
UPDATE_VOLUME( volume_right, rvol ); \
); \
}
#define MONO_DEST_MIX_LINEAR(op, upd, o0, o1) { \
*dst++ op MULSC(LINEAR(x[o0], x[o1]), vol); \
@ -208,16 +253,51 @@ DUMB_RESAMPLER *dumb_start_resampler(SRCTYPE *src, int src_channels, int32 pos,
if ( volume_right ) volume_right->volume = (float)rvolr / 16777216.0f; \
}
#define MONO_DEST_VOLUMES_ARE_ZERO (lvol == 0 && lvolt == 0 && rvol == 0 && rvolt == 0)
#define MONO_DEST_MIX_ALIAS(op, upd, offset) { \
*dst++ op ALIAS(x[(offset)*2], lvol) + ALIAS(x[(offset)*2+1], rvol); \
if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
#define POKE_ALIAS(offset) { \
int deltal = ALIAS(x[(offset)*2+0]) - resampler->last_amp[0]; \
int deltar = ALIAS(x[(offset)*2+1]) - resampler->last_amp[1]; \
resampler->last_amp[0] += deltal; \
resampler->last_amp[1] += deltar; \
if ( deltal ) blip_add_delta( resampler->blip_buffer[0], resampler->last_clock, deltal ); \
if ( deltar ) blip_add_delta( resampler->blip_buffer[1], resampler->last_clock, deltar ); \
resampler->last_clock += inv_dt; \
}
#define STEREO_DEST_MIX_ALIAS(op, upd, offset) { \
*dst++ op ALIAS(x[(offset)*2], lvol); \
*dst++ op ALIAS(x[(offset)*2+1], rvol); \
if ( upd ) UPDATE_VOLUME( volume_left, lvol ); \
if ( upd ) UPDATE_VOLUME( volume_right, rvol ); \
#define MONO_DEST_PEEK_ALIAS { \
*dst = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), lvol ) + \
MULSC( blip_peek_sample( resampler->blip_buffer[1] ), rvol ); \
}
#define MONO_DEST_MIX_ALIAS(count) { \
int n = 0; \
resampler->last_clock -= count * 65536; \
blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
blip_end_frame( resampler->blip_buffer[1], count * 65536 ); \
blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
blip_read_samples( resampler->blip_buffer[1], blip_samples + 256, count ); \
LOOP4( count, \
*dst++ += MULSC( blip_samples[n], lvol ) + MULSC( blip_samples[256+n], rvol ); \
n++; \
UPDATE_VOLUME( volume_left, lvol ); \
UPDATE_VOLUME( volume_right, rvol ); \
); \
}
#define STEREO_DEST_PEEK_ALIAS { \
*dst++ = MULSC( blip_peek_sample( resampler->blip_buffer[0] ), lvol ); \
*dst++ = MULSC( blip_peek_sample( resampler->blip_buffer[1] ), rvol ); \
}
#define STEREO_DEST_MIX_ALIAS(count) { \
int n = 0; \
resampler->last_clock -= count * 65536; \
blip_end_frame( resampler->blip_buffer[0], count * 65536 ); \
blip_end_frame( resampler->blip_buffer[1], count * 65536 ); \
blip_read_samples( resampler->blip_buffer[0], blip_samples, count ); \
blip_read_samples( resampler->blip_buffer[1], blip_samples + 256, count ); \
LOOP4( count, \
*dst++ += MULSC( blip_samples[n], lvol); \
*dst++ += MULSC( blip_samples[256+n], rvol); \
n++; \
UPDATE_VOLUME( volume_left, lvol ); \
UPDATE_VOLUME( volume_right, rvol ); \
); \
}
#define MONO_DEST_MIX_LINEAR(op, upd, o0, o1) { \
*dst++ op MULSC(LINEAR(x[(o0)*2], x[(o1)*2]), lvol) + MULSC(LINEAR(x[(o0)*2+1], x[(o1)*2+1]), rvol); \

View file

@ -29,15 +29,38 @@
#define END_RAMPING
#define RAMP_DOWN
static IT_PLAYING *alloc_playing(DUMB_IT_SIGRENDERER *itsr)
static IT_PLAYING *new_playing(DUMB_IT_SIGRENDERER *itsr)
{
IT_PLAYING *r;
if (itsr->free_playing != NULL)
{
IT_PLAYING *pl = itsr->free_playing;
itsr->free_playing = pl->next;
return pl;
r = itsr->free_playing;
itsr->free_playing = r->next;
blip_clear(r->resampler.blip_buffer[0]);
blip_clear(r->resampler.blip_buffer[1]);
return r;
}
return (IT_PLAYING *)malloc(sizeof(IT_PLAYING));
r = (IT_PLAYING *)malloc(sizeof(IT_PLAYING));
if (r)
{
r->resampler.blip_buffer[0] = blip_new( 256 );
if ( !r->resampler.blip_buffer[0] )
{
free( r );
return NULL;
}
r->resampler.blip_buffer[1] = blip_new( 256 );
if ( !r->resampler.blip_buffer[1] )
{
free( r->resampler.blip_buffer[0] );
free( r );
return NULL;
}
blip_set_rates(r->resampler.blip_buffer[0], 65536, 1);
blip_set_rates(r->resampler.blip_buffer[1], 65536, 1);
}
return r;
}
static void free_playing(DUMB_IT_SIGRENDERER *itsr, IT_PLAYING *playing)
@ -46,6 +69,13 @@ static void free_playing(DUMB_IT_SIGRENDERER *itsr, IT_PLAYING *playing)
itsr->free_playing = playing;
}
static void free_playing_orig(IT_PLAYING * r)
{
blip_delete( r->resampler.blip_buffer[1] );
blip_delete( r->resampler.blip_buffer[0] );
free( r );
}
static IT_PLAYING *dup_playing(IT_PLAYING *src, IT_CHANNEL *dstchannel, IT_CHANNEL *srcchannel)
{
IT_PLAYING *dst;
@ -135,6 +165,19 @@ static IT_PLAYING *dup_playing(IT_PLAYING *src, IT_CHANNEL *dstchannel, IT_CHANN
dst->resampler = src->resampler;
dst->resampler.pickup_data = dst;
dst->resampler.blip_buffer[0] = blip_dup( dst->resampler.blip_buffer[0] );
if ( !dst->resampler.blip_buffer[0] )
{
free( dst );
return NULL;
}
dst->resampler.blip_buffer[1] = blip_dup( dst->resampler.blip_buffer[1] );
if ( !dst->resampler.blip_buffer[1] )
{
blip_delete( dst->resampler.blip_buffer[0] );
free( dst );
return NULL;
}
dst->time_lost = src->time_lost;
//dst->output = src->output;
@ -1582,7 +1625,7 @@ static void it_retrigger_note(DUMB_IT_SIGRENDERER *sigrenderer, IT_CHANNEL *chan
if (channel->playing)
free_playing(sigrenderer, channel->playing);
channel->playing = alloc_playing(sigrenderer);
channel->playing = new_playing(sigrenderer);
if (!channel->playing)
return;
@ -2889,9 +2932,7 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
{
DUMB_IT_SIGDATA *sigdata = sigrenderer->sigdata;
IT_CHANNEL *channel = &sigrenderer->channel[(int)entry->channel];
IT_PLAYING playing;
playing.sample = 0;
IT_PLAYING * playing = NULL;
if (entry->mask & IT_ENTRY_INSTRUMENT) {
int oldsample = channel->sample;
@ -2901,7 +2942,8 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
if (channel->playing &&
!((entry->mask & IT_ENTRY_NOTE) && entry->note >= 120) &&
!((entry->mask & IT_ENTRY_EFFECT) && entry->effect == IT_XM_KEY_OFF && entry->effectvalue == 0)) {
playing = *channel->playing;
playing = dup_playing(channel->playing, channel, channel);
if (!playing) return;
if (!(sigdata->flags & IT_WAS_A_MOD)) {
/* Retrigger vol/pan envelopes if enabled, and cancel fadeout.
* Also reset vol/pan to that of _original_ instrument.
@ -2934,12 +2976,11 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
channel->playing = NULL;
}
} else {
if (!channel->playing) {
channel->playing = alloc_playing(sigrenderer);
if (!channel->playing) return;
if (channel->playing) {
free_playing(sigrenderer, channel->playing);
}
*channel->playing = playing;
playing.sample = (IT_SAMPLE *)-1;
channel->playing = playing;
playing = NULL;
channel->playing->declick_stage = 0;
channel->playing->declick_volume = 1.f / 256.f;
#else
@ -2985,7 +3026,11 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
if (channel->playing) {
#ifdef RAMP_DOWN
int i;
if (playing.sample) *channel->playing = playing;
if (playing) {
free_playing(sigrenderer, channel->playing);
channel->playing = playing;
playing = NULL;
}
for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++) {
if (!sigrenderer->playing[i]) {
channel->playing->declick_stage = 2;
@ -3003,6 +3048,7 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
channel->playing = NULL;
#endif
}
if (playing) free_playing(sigrenderer, playing);
return;
} else if (channel->playing && (entry->mask & IT_ENTRY_VOLPAN) && ((entry->volpan>>4) == 0xF)) {
/* Don't retrigger note; portamento in the volume column. */
@ -3015,20 +3061,26 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
channel->destnote = IT_NOTE_OFF;
if (!channel->playing) {
channel->playing = alloc_playing(sigrenderer);
if (!channel->playing)
channel->playing = new_playing(sigrenderer);
if (!channel->playing) {
if (playing) free_playing(sigrenderer, playing);
return;
}
// Adding the following seems to do the trick for the case where a piece starts with an instrument alone and then some notes alone.
retrigger_xm_envelopes(channel->playing);
}
#ifdef RAMP_DOWN
else if (playing.sample != (IT_SAMPLE *)-1) {
else if (playing) {
/* volume rampy stuff! move note to NNA */
int i;
IT_PLAYING * ptemp = alloc_playing(sigrenderer);
if (!ptemp) return;
if (playing.sample) *ptemp = playing;
else *ptemp = *channel->playing;
IT_PLAYING * ptemp;
if (playing->sample) ptemp = playing;
else ptemp = channel->playing;
if (!ptemp) {
if (playing) free_playing(sigrenderer, playing);
return;
}
playing = NULL;
for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++) {
if (!sigrenderer->playing[i]) {
ptemp->declick_stage = 2;
@ -3189,6 +3241,8 @@ static void process_xm_note_data(DUMB_IT_SIGRENDERER *sigrenderer, IT_ENTRY *ent
break;
}
}
if (playing) free_playing(sigrenderer, playing);
}
@ -5165,7 +5219,7 @@ void _dumb_it_end_sigrenderer(sigrenderer_t *vsigrenderer)
for (i = 0; i < DUMB_IT_N_CHANNELS; i++) {
if (sigrenderer->channel[i].playing)
free(sigrenderer->channel[i].playing);
free_playing_orig(sigrenderer->channel[i].playing);
#ifdef BIT_ARRAY_BULLSHIT
bit_array_destroy(sigrenderer->channel[i].played_patjump);
#endif
@ -5173,12 +5227,12 @@ void _dumb_it_end_sigrenderer(sigrenderer_t *vsigrenderer)
for (i = 0; i < DUMB_IT_N_NNA_CHANNELS; i++)
if (sigrenderer->playing[i])
free(sigrenderer->playing[i]);
free_playing_orig(sigrenderer->playing[i]);
for (playing = sigrenderer->free_playing; playing != NULL; playing = next)
{
next = playing->next;
free(playing);
free_playing_orig(playing);
}
dumb_destroy_click_remover_array(sigrenderer->n_channels, sigrenderer->click_remover);

View file

@ -877,6 +877,10 @@
RelativePath="..\..\src\helpers\barray.c"
>
</File>
<File
RelativePath="..\..\src\helpers\blip_buf.c"
>
</File>
<File
RelativePath="..\..\src\helpers\clickrem.c"
>
@ -1960,6 +1964,10 @@
<Filter
Name="internal"
>
<File
RelativePath="..\..\include\internal\blip_buf.h"
>
</File>
<File
RelativePath="..\..\include\internal\dumb.h"
>