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sound, improved and optimized reimplementation of libretro lowpass filter

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This commit is contained in:
kub 2020-12-21 23:22:00 +01:00
parent b437951ade
commit 30969671e5
5 changed files with 141 additions and 123 deletions
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55
pico/sound/mix.c
View file

@ -1,6 +1,7 @@
/*
* some code for sample mixing
* (C) notaz, 2006,2007
* (C) kub, 2019,2020 added filtering
*
* This work is licensed under the terms of MAME license.
* See COPYING file in the top-level directory.
@ -13,78 +14,97 @@
/* limitter */
#define Limit16(val) \
val -= val >> 2; /* reduce level to avoid clipping */ \
if ((short)val != val) val = (val < 0 ? MINOUT : MAXOUT)
int mix_32_to_16l_level;
static struct iir2 { // 2-pole IIR
int x[2]; // sample buffer
static struct iir {
int alpha; // alpha for EMA low pass
int y[2]; // filter intermediates
int i;
} lfi2, rfi2;
// NB ">>" rounds to -infinity, "/" to 0. To compensate the effect possibly use
// "-(-y>>n)" (round to +infinity) instead of "y>>n" in places.
// NB uses Q12 fixpoint; samples mustn't have more than 20 bits for this.
// NB uses fixpoint; samples mustn't have more than (32-QB) bits. Adding the
// outputs of the sound sources together yields a max. of 18 bits, restricting
// QB to a maximum of 14.
#define QB 12
// NB alpha for DC filtering shouldn't be smaller than 1/(1<<QB) to avoid loss.
// exponential moving average combined DC filter and lowpass filter
// y0[n] = (x[n]-y0[n-1])*alpha+y0[n-1], y1[n] = (y0[n] - y1[n-1])*(1-1/8192)
static inline int filter_band(struct iir *fi2, int x)
{
// low pass. alpha is Q8 to avoid loss by 32 bit overflow.
// fi2->y[0] += ((x<<(QB-8)) - (fi2->y[0]>>8)) * fi2->alpha;
fi2->y[0] += (x - (fi2->y[0]>>QB)) * fi2->alpha;
// DC filter. for alpha=1-1/8192 cutoff ~1HZ, for 1-1/1024 ~7Hz
fi2->y[1] += (fi2->y[0] - fi2->y[1]) >> QB;
return (fi2->y[0] - fi2->y[1]) >> QB;
}
// exponential moving average filter for DC filtering
// y[n] = (x[n]-y[n-1])*(1/8192) (corner approx. 20Hz, gain 1)
static inline int filter_exp(struct iir2 *fi2, int x)
// y[n] = (x[n]-y[n-1])*(1-1/8192) (corner approx. 1Hz, gain 1)
static inline int filter_exp(struct iir *fi2, int x)
{
int xf = (x<<QB) - fi2->y[0];
fi2->y[0] += xf >> 13;
xf -= xf >> 2; // level reduction to avoid clipping from overshoot
return xf>>QB;
fi2->y[1] += ((x << QB) - fi2->y[1]) >> QB;
return x - (fi2->y[1] >> QB);
}
// unfiltered (for testing)
static inline int filter_null(struct iir2 *fi2, int x)
static inline int filter_null(struct iir *fi2, int x)
{
return x;
}
#define filter filter_band
#define mix_32_to_16l_stereo_core(dest, src, count, lv, fl) { \
int l, r; \
struct iir lf = lfi2, rf = rfi2; \
\
for (; count > 0; count--) \
{ \
l = r = *dest; \
l += *src++ >> lv; \
r += *src++ >> lv; \
l = fl(&lfi2, l); \
r = fl(&rfi2, r); \
l = fl(&lf, l); \
r = fl(&rf, r); \
Limit16(l); \
Limit16(r); \
*dest++ = l; \
*dest++ = r; \
} \
lfi2 = lf, rfi2 = rf; \
}
void mix_32_to_16l_stereo_lvl(short *dest, int *src, int count)
{
mix_32_to_16l_stereo_core(dest, src, count, mix_32_to_16l_level, filter_exp);
mix_32_to_16l_stereo_core(dest, src, count, mix_32_to_16l_level, filter);
}
void mix_32_to_16l_stereo(short *dest, int *src, int count)
{
mix_32_to_16l_stereo_core(dest, src, count, 0, filter_exp);
mix_32_to_16l_stereo_core(dest, src, count, 0, filter);
}
void mix_32_to_16_mono(short *dest, int *src, int count)
{
int l;
struct iir lf = lfi2;
for (; count > 0; count--)
{
l = *dest;
l += *src++;
l = filter_exp(&lfi2, l);
l = filter(&lf, l);
Limit16(l);
*dest++ = l;
}
lfi2 = lf;
}
@ -118,8 +138,9 @@ void mix_16h_to_32_s2(int *dest_buf, short *mp3_buf, int count)
}
}
void mix_reset(void)
void mix_reset(int alpha_q16)
{
memset(&lfi2, 0, sizeof(lfi2));
memset(&rfi2, 0, sizeof(rfi2));
lfi2.alpha = rfi2.alpha = (0x10000-alpha_q16) >> 4; // filter alpha, Q12
}

2
pico/sound/mix.h
View file

@ -8,4 +8,4 @@ void mix_32_to_16_mono(short *dest, int *src, int count);
extern int mix_32_to_16l_level;
void mix_32_to_16l_stereo_lvl(short *dest, int *src, int count);
void mix_reset(void);
void mix_reset(int alpha_q16);

117
pico/sound/mix_arm.S
View file

@ -154,34 +154,46 @@ m16_32_s2_no_unal2:
@ limit
@ reg=int_sample, lr=1, r3=tmp, kills flags
@ reg=int_sample, r12=1, r8=tmp, kills flags
.macro Limit reg
add r3, lr, \reg, asr #15
bics r3, r3, #1 @ in non-overflow conditions r3 is 0 or 1
sub \reg, \reg, \reg, asr #2 @ reduce audio lvl some to avoid clipping
add r8, r12, \reg, asr #15
bics r8, r8, #1 @ in non-overflow conditions r8 is 0 or 1
movne \reg, #0x8000
subpl \reg, \reg, #1
.endm
@ limit and shift up by 16
@ reg=int_sample, lr=1, r3=tmp, kills flags
@ reg=int_sample, r12=1, r8=tmp, kills flags
.macro Limitsh reg
add r3, lr, \reg, asr #15
bics r3, r3, #1 @ in non-overflow conditions r3 is 0 or 1
sub \reg, \reg, \reg, asr #2 @ reduce audio lvl some to avoid clipping
add r8, r12,\reg, asr #15
bics r8, r8, #1 @ in non-overflow conditions r8 is 0 or 1
moveq \reg, \reg, lsl #16
movne \reg, #0x80000000
subpl \reg, \reg, #0x00010000
.endm
@ filter out DC offset
@ in=int_sample (max 20 bit), y=filter memory, r3=tmp
@ in=int_sample (max 20 bit), y=filter memory, r8=tmp
.macro DCfilt in y
rsb r3, \y, \in, lsl #12 @ fixpoint 20.12
add \y, \y, r3, asr #13
sub r3, r3, r3, asr #2 @ reduce audio lvl some
asr \in, r3, #12
rsb r8, \y, \in, lsl #12 @ fixpoint 20.12
add \y, \y, r8, asr #12 @ alpha = 1-1/4094
sub \in, \in, \y, asr #12
.endm
@ lowpass filter
@ in=int_sample (max 20 bit), y=filter memory, r12=alpha(Q8), r8=tmp
.macro LPfilt in y
@ asr r8, \y, #8
@ rsb r8, r8, \in, lsl #4 @ fixpoint 20.12
sub r8, \in, \y, asr #12 @ fixpoint 20.12
mla \y, r8, r12, \y
asr \in, \y, #12
.endm
@ mix 32bit audio (with 16bits really used, upper bits indicate overflow) with normal 16 bit audio with left channel only
@ warning: this function assumes dest is word aligned
.global mix_32_to_16l_stereo @ short *dest, int *src, int count
@ -193,9 +205,10 @@ mix_32_to_16l_stereo:
subs r2, r2, #4
bmi m32_16l_st_end
mov lr, #1
ldr r12, =filter
ldmia r12, {r10-r11}
ldr r8, [r12], #4
ldmia r12, {r3,r10-r11,lr}
str r8, [sp, #-4]!
m32_16l_st_loop:
ldmia r0, {r8,r12}
@ -206,10 +219,16 @@ m32_16l_st_loop:
add r5, r5, r8, asr #16
add r6, r6, r12,asr #16
add r7, r7, r12,asr #16
ldr r12,[sp]
LPfilt r4, r3
LPfilt r5, lr
LPfilt r6, r3
LPfilt r7, lr
DCfilt r4, r10
DCfilt r5, r11
DCfilt r6, r10
DCfilt r7, r11
mov r12,#1
Limitsh r4
Limitsh r5
Limitsh r6
@ -228,8 +247,12 @@ m32_16l_st_end:
ldmia r1!,{r4,r5}
add r4, r4, r6
add r5, r5, r6
ldr r12,[sp]
LPfilt r4, r3
LPfilt r5, lr
DCfilt r4, r10
DCfilt r5, r11
mov r12,#1
Limitsh r4
Limitsh r5
orr r4, r5, r4, lsr #16
@ -237,7 +260,9 @@ m32_16l_st_end:
m32_16l_st_no_unal2:
ldr r12, =filter
stmia r12, {r10-r11}
add r12,r12, #4
stmia r12, {r3,r10-r11,lr}
add sp, sp, #4
ldmfd sp!, {r4-r8,r10-r11,lr}
bx lr
@ -248,9 +273,10 @@ m32_16l_st_no_unal2:
mix_32_to_16_mono:
stmfd sp!, {r4-r8,r10-r11,lr}
mov lr, #1
ldr r12, =filter
ldr r10, [r12]
ldr r8, [r12], #4
ldmia r12, {r10-r11}
str r8, [sp, #-4]!
@ check if dest is word aligned
tst r0, #2
@ -259,6 +285,10 @@ mix_32_to_16_mono:
ldr r4, [r1], #4
sub r2, r2, #1
add r4, r4, r5
ldr r12,[sp]
LPfilt r4, r11
DCfilt r4, r10
mov r12,#1
Limit r4
strh r4, [r0], #2
@ -275,10 +305,16 @@ m32_16_mo_loop:
add r7, r7, r12,asr #16
mov r12,r12,lsl #16
add r6, r6, r12,asr #16
ldr r12,[sp]
LPfilt r4, r11
LPfilt r5, r11
LPfilt r6, r11
LPfilt r7, r11
DCfilt r4, r10
DCfilt r5, r10
DCfilt r6, r10
DCfilt r7, r10
mov r12,#1
Limitsh r4
Limitsh r5
Limitsh r6
@ -298,8 +334,12 @@ m32_16_mo_end:
add r5, r5, r6, asr #16
mov r6, r6, lsl #16
add r4, r4, r6, asr #16
ldr r12,[sp]
LPfilt r4, r11
LPfilt r5, r11
DCfilt r4, r10
DCfilt r5, r10
mov r12,#1
Limitsh r4
Limitsh r5
orr r4, r5, r4, lsr #16
@ -311,13 +351,18 @@ m32_16_mo_no_unal2:
ldrsh r5, [r0]
ldr r4, [r1], #4
add r4, r4, r5
ldr r12,[sp]
LPfilt r4, r11
DCfilt r4, r10
mov r12,#1
Limit r4
strh r4, [r0], #2
m32_16_mo_no_unal:
ldr r12, =filter
str r10, [r12]
add r12,r12, #4
stmia r12, {r10-r11}
add sp, sp, #4
ldmfd sp!, {r4-r8,r10-r11,lr}
bx lr
@ -344,7 +389,9 @@ mix_32_to_16l_stereo_lvl:
mov lr, #1
ldr r9, [r9]
ldr r12, =filter
ldm r12, {r10-r11}
ldr r8, [r12], #4
ldmia r12, {r3,r10-r11,lr}
str r8, [sp, #-4]!
mov r2, r2, lsl #1
subs r2, r2, #4
@ -363,10 +410,16 @@ m32_16l_st_l_loop:
mov r5, r5, asr r9
mov r6, r6, asr r9
mov r7, r7, asr r9
ldr r12,[sp]
LPfilt r4, r3
LPfilt r5, lr
LPfilt r6, r3
LPfilt r7, lr
DCfilt r4, r10
DCfilt r5, r11
DCfilt r6, r10
DCfilt r7, r11
mov r12,#1
Limitsh r4
Limitsh r5
Limitsh r6
@ -387,8 +440,12 @@ m32_16l_st_l_end:
add r5, r5, r6
mov r4, r4, asr r9
mov r5, r5, asr r9
ldr r12,[sp]
LPfilt r4, r3
LPfilt r5, lr
DCfilt r4, r10
DCfilt r5, r11
mov r12,#1
Limitsh r4
Limitsh r5
orr r4, r5, r4, lsr #16
@ -396,22 +453,32 @@ m32_16l_st_l_end:
m32_16l_st_l_no_unal2:
ldr r12, =filter
stmia r12, {r10-r11}
add r12,r12, #4
stmia r12, {r3,r10-r11,lr}
add sp, sp, #4
ldmfd sp!, {r4-r11,lr}
bx lr
#endif /* __GP2X__ */
.global mix_reset @ void
.global mix_reset @ int alpha_q16
mix_reset:
ldr r0, =filter
ldr r2, =filter
rsb r0, r0, #0x10000
@ asr r0, r0, #8
asr r0, r0, #4
str r0, [r2], #4
mov r1, #0
str r1, [r0]
str r1, [r0, #4]
str r1, [r2], #4
str r1, [r2], #4
str r1, [r2], #4
str r1, [r2], #4
bx lr
.data
filter:
.ds 8
.ds 4 @ alpha_q8
.ds 8 @ filter history for left channel
.ds 8 @ filter history for right channel
@ vim:filetype=armasm

81
pico/sound/sound.c
View file

@ -26,73 +26,6 @@ short cdda_out_buffer[2*1152];
// sn76496
extern int *sn76496_regs;
// Low pass filter 'previous' samples
static int32_t lpf_lp;
static int32_t lpf_rp;
static void low_pass_filter_stereo(int *buf32, int length)
{
int samples = length;
int *out32 = buf32;
// Restore previous samples
int32_t lpf_l = lpf_lp;
int32_t lpf_r = lpf_rp;
// Single-pole low-pass filter (6 dB/octave)
int32_t factor_a = PicoIn.sndFilterRange;
int32_t factor_b = 0x10000 - factor_a;
do
{
// Apply low-pass filter
lpf_l = (lpf_l * factor_a) + (out32[0] * factor_b);
lpf_r = (lpf_r * factor_a) + (out32[1] * factor_b);
// 16.16 fixed point
lpf_l >>= 16;
lpf_r >>= 16;
// Update sound buffer
*out32++ = lpf_l;
*out32++ = lpf_r;
}
while (--samples);
// Save last samples for next frame
lpf_lp = lpf_l;
lpf_rp = lpf_r;
}
static void low_pass_filter_mono(int *buf32, int length)
{
int samples = length;
int *out32 = buf32;
// Restore previous sample
int32_t lpf_l = lpf_lp;
// Single-pole low-pass filter (6 dB/octave)
int32_t factor_a = PicoIn.sndFilterRange;
int32_t factor_b = 0x10000 - factor_a;
do
{
// Apply low-pass filter
lpf_l = (lpf_l * factor_a) + (out32[0] * factor_b);
// 16.16 fixed point
lpf_l >>= 16;
// Update sound buffer
*out32++ = lpf_l;
}
while (--samples);
// Save last sample for next frame
lpf_lp = lpf_l;
}
void (*low_pass_filter)(int *buf32, int length) = low_pass_filter_stereo;
// ym2413
#define YM2413_CLK 3579545
OPLL old_opll;
@ -119,11 +52,7 @@ PICO_INTERNAL void PsndReset(void)
PsndRerate(0);
timers_reset();
// Reset low pass filter
lpf_lp = 0;
lpf_rp = 0;
mix_reset();
mix_reset(PicoIn.sndFilter ? PicoIn.sndFilterRange : 0);
}
@ -179,9 +108,6 @@ void PsndRerate(int preserve_state)
// set mixer
PsndMix_32_to_16l = (PicoIn.opt & POPT_EN_STEREO) ? mix_32_to_16l_stereo : mix_32_to_16_mono;
// set low pass filter
low_pass_filter = (PicoIn.opt & POPT_EN_STEREO) ? low_pass_filter_stereo : low_pass_filter_mono;
if (PicoIn.AHW & PAHW_PICO)
PicoReratePico();
}
@ -463,11 +389,6 @@ static int PsndRender(int offset, int length)
if ((PicoIn.AHW & PAHW_32X) && (PicoIn.opt & POPT_EN_PWM))
p32x_pwm_update(buf32, length-offset, stereo);
// Apply low pass filter, if required
if (PicoIn.sndFilter == 1) {
low_pass_filter(buf32, length);
}
// convert + limit to normal 16bit output
PsndMix_32_to_16l(PicoIn.sndOut+(offset<<stereo), buf32, length-offset);

9
platform/libretro/libretro.c
View file

@ -70,6 +70,7 @@ int _newlib_vm_size_user = 1 << TARGET_SIZE_2;
#include <pico/pico_int.h>
#include <pico/state.h>
#include <pico/patch.h>
#include <pico/sound/mix.h>
#include "../common/input_pico.h"
#include "../common/version.h"
#include <libretro.h>
@ -1434,6 +1435,8 @@ static void update_variables(bool first_run)
unsigned old_frameskip_type;
int old_vout_format;
double new_sound_rate;
unsigned short old_snd_filter;
int32_t old_snd_filter_range;
var.value = NULL;
var.key = "picodrive_input1";
@ -1539,6 +1542,7 @@ static void update_variables(bool first_run)
PicoIn.opt &= ~POPT_EN_DRC;
#endif
old_snd_filter = PicoIn.sndFilter;
var.value = NULL;
var.key = "picodrive_audio_filter";
PicoIn.sndFilter = 0;
@ -1547,6 +1551,7 @@ static void update_variables(bool first_run)
PicoIn.sndFilter = 1;
}
old_snd_filter_range = PicoIn.sndFilterRange;
var.value = NULL;
var.key = "picodrive_lowpass_range";
PicoIn.sndFilterRange = (60 * 65536) / 100;
@ -1554,6 +1559,10 @@ static void update_variables(bool first_run)
PicoIn.sndFilterRange = (atoi(var.value) * 65536) / 100;
}
if (old_snd_filter != PicoIn.sndFilter || old_snd_filter_range != PicoIn.sndFilterRange) {
mix_reset(PicoIn.sndFilter ? PicoIn.sndFilterRange : 0);
}
old_frameskip_type = frameskip_type;
frameskip_type = 0;
var.key = "picodrive_frameskip";