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RetroArch/audio/drivers_resampler/cc_resampler.c
twinaphex 4a818c24ea (cc_resampler.c) Simplifications
2015-09-12 22:11:02 +02:00

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/* RetroArch - A frontend for libretro.
* Copyright (C) 2014-2015 - Ali Bouhlel ( aliaspider@gmail.com )
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see <http://www.gnu.org/licenses/>.
*/
/* Convoluted Cosine Resampler */
#include <math.h>
#include <stdint.h>
#include <stdlib.h>
#ifdef __SSE__
#include <xmmintrin.h>
#endif
#include <retro_inline.h>
#include <memalign.h>
#include "../audio_resampler_driver.h"
/* Since SSE and NEON don't provide support for trigonometric functions
* we approximate those with polynoms
*
* CC_RESAMPLER_PRECISION defines how accurate the approximation is
* a setting of 5 or more means full precison.
* setting 0 doesn't use a polynom
* setting 1 uses P(X) = X - (3/4)*X^3 + (1/4)*X^5
*
* only 0 and 1 are implemented for SSE and NEON currently
*
* the MIPS_ARCH_ALLEGREX target doesnt require this setting since it has
* native support for the required functions so it will always use full precision.
*/
#ifndef CC_RESAMPLER_PRECISION
#define CC_RESAMPLER_PRECISION 1
#endif
#ifndef min
#define min(a, b) ((a) < (b) ? (a) : (b))
#endif
typedef struct rarch_CC_resampler
{
audio_frame_float_t buffer[4];
float distance;
void (*process)(void *re, struct resampler_data *data);
} rarch_CC_resampler_t;
#ifdef _MIPS_ARCH_ALLEGREX
static void resampler_CC_process(void *re_, struct resampler_data *data)
{
float ratio, fraction;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = (audio_frame_float_t*)
(inp + data->input_frames);
audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
(void)re_;
__asm__ (
".set push\n"
".set noreorder\n"
"mtv %2, s700 \n" /* 700 = data->ratio = b */
/* "vsat0.s s700, s700 \n" */
"vrcp.s s701, s700 \n" /* 701 = 1.0 / b */
"vadd.s s702, s700, s700 \n" /* 702 = 2 * b */
"vmul.s s703, s700, s710 \n" /* 703 = b * pi */
"mfv %0, s701 \n"
"mfv %1, s730 \n"
".set pop\n"
: "=r"(ratio), "=r"(fraction)
: "r"((float)data->ratio)
);
for (;;)
{
while (fraction < ratio)
{
if (inp == inp_max)
goto done;
__asm__ (
".set push \n"
".set noreorder \n"
"lv.s s620, 0(%1) \n"
"lv.s s621, 4(%1) \n"
"vsub.s s731, s701, s730 \n"
"vadd.q c600, c730[-X,Y,-X,Y], c730[1/2,1/2,-1/2,-1/2]\n"
"vmul.q c610, c600, c700[Z,Z,Z,Z] \n" /* *2*b */
"vmul.q c600, c600, c700[W,W,W,W] \n" /* *b*pi */
"vsin.q c610, c610 \n"
"vadd.q c600, c600, c610 \n"
"vmul.q c600[-1:1,-1:1,-1:1,-1:1], c600, c710[Y,Y,Y,Y] \n"
"vsub.p c600, c600, c602 \n"
"vmul.q c620, c620[X,Y,X,Y], c600[X,X,Y,Y] \n"
"vadd.q c720, c720, c620 \n"
"vadd.s s730, s730, s730[1] \n"
"mfv %0, s730 \n"
".set pop \n"
: "=r"(fraction)
: "r"(inp));
inp++;
}
__asm__ (
".set push \n"
".set noreorder \n"
"vmul.p c720, c720, c720[1/2,1/2] \n"
"sv.s s720, 0(%1) \n"
"sv.s s721, 4(%1) \n"
"vmov.q c720, c720[Z,W,0,0] \n"
"vsub.s s730, s730, s701 \n"
"mfv %0, s730 \n"
".set pop \n"
: "=r"(fraction)
: "r"(outp));
outp++;
}
/* The VFPU state is assumed to remain intact
* in-between calls to resampler_CC_process. */
done:
data->output_frames = outp - (audio_frame_float_t*)data->data_out;
}
static void *resampler_CC_init(const struct resampler_config *config,
double bandwidth_mod, resampler_simd_mask_t mask)
{
(void)mask;
(void)bandwidth_mod;
(void)config;
__asm__ (
".set push\n"
".set noreorder\n"
"vcst.s s710, VFPU_PI \n" /* 710 = pi */
"vcst.s s711, VFPU_1_PI \n" /* 711 = 1.0 / (pi) */
"vzero.q c720 \n"
"vzero.q c730 \n"
".set pop\n");
return (void*)-1;
}
#else
#if defined(__SSE__)
#define CC_RESAMPLER_IDENT "SSE"
static void resampler_CC_downsample(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = (audio_frame_float_t*)(inp + data->input_frames);
audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
float ratio = 1.0 / data->ratio;
float b = data->ratio; /* cutoff frequency. */
__m128 vec_previous = _mm_loadu_ps((float*)&re->buffer[0]);
__m128 vec_current = _mm_loadu_ps((float*)&re->buffer[2]);
while (inp != inp_max)
{
__m128 vec_ww1, vec_ww2;
__m128 vec_w_previous;
__m128 vec_w_current;
__m128 vec_in;
__m128 vec_ratio =
_mm_mul_ps(_mm_set_ps1(ratio), _mm_set_ps(3.0, 2.0, 1.0, 0.0));
__m128 vec_w = _mm_sub_ps(_mm_set_ps1(re->distance), vec_ratio);
__m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5));
__m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5));
__m128 vec_b = _mm_set_ps1(b);
vec_w1 = _mm_mul_ps(vec_w1, vec_b);
vec_w2 = _mm_mul_ps(vec_w2, vec_b);
(void)vec_ww1;
(void)vec_ww2;
#if (CC_RESAMPLER_PRECISION > 0)
vec_ww1 = _mm_mul_ps(vec_w1, vec_w1);
vec_ww2 = _mm_mul_ps(vec_w2, vec_w2);
vec_ww1 = _mm_mul_ps(vec_ww1, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww1));
vec_ww2 = _mm_mul_ps(vec_ww2, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww2));
vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww1);
vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww2);
vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1));
vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2));
#endif
vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5));
vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5));
vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5));
vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5));
vec_w = _mm_sub_ps(vec_w1, vec_w2);
vec_w_previous =
_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0));
vec_w_current =
_mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2));
vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp);
vec_in = _mm_shuffle_ps(vec_in,vec_in,_MM_SHUFFLE(1, 0, 1, 0));
vec_previous =
_mm_add_ps(vec_previous, _mm_mul_ps(vec_in, vec_w_previous));
vec_current =
_mm_add_ps(vec_current, _mm_mul_ps(vec_in, vec_w_current));
re->distance++;
inp++;
if (re->distance > (ratio + 0.5))
{
_mm_storel_pi((__m64*)outp, vec_previous);
vec_previous =
_mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2));
vec_current =
_mm_shuffle_ps(vec_current,_mm_setzero_ps(),_MM_SHUFFLE(1, 0, 3, 2));
re->distance -= ratio;
outp++;
}
}
_mm_storeu_ps((float*)&re->buffer[0], vec_previous);
_mm_storeu_ps((float*)&re->buffer[2], vec_current);
data->output_frames = outp - (audio_frame_float_t*)data->data_out;
}
static void resampler_CC_upsample(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = (audio_frame_float_t*)(inp + data->input_frames);
audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
float b = min(data->ratio, 1.00); /* cutoff frequency. */
float ratio = 1.0 / data->ratio;
__m128 vec_previous = _mm_loadu_ps((float*)&re->buffer[0]);
__m128 vec_current = _mm_loadu_ps((float*)&re->buffer[2]);
while (inp != inp_max)
{
__m128 vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp);
vec_previous =
_mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2));
vec_current =
_mm_shuffle_ps(vec_current,vec_in,_MM_SHUFFLE(1, 0, 3, 2));
while (re->distance < 1.0)
{
__m128 vec_w_previous, vec_w_current, vec_out;
#if (CC_RESAMPLER_PRECISION > 0)
__m128 vec_ww1, vec_ww2;
#endif
__m128 vec_w =
_mm_add_ps(_mm_set_ps1(re->distance), _mm_set_ps(-2.0, -1.0, 0.0, 1.0));
__m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5));
__m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5));
__m128 vec_b = _mm_set_ps1(b);
vec_w1 = _mm_mul_ps(vec_w1, vec_b);
vec_w2 = _mm_mul_ps(vec_w2, vec_b);
#if (CC_RESAMPLER_PRECISION > 0)
vec_ww1 = _mm_mul_ps(vec_w1, vec_w1);
vec_ww2 = _mm_mul_ps(vec_w2, vec_w2);
vec_ww1 = _mm_mul_ps(vec_ww1,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww1));
vec_ww2 = _mm_mul_ps(vec_ww2,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww2));
vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww1);
vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww2);
vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1));
vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2));
#endif
vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5));
vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5));
vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5));
vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5));
vec_w = _mm_sub_ps(vec_w1, vec_w2);
vec_w_previous = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0));
vec_w_current = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2));
vec_out = _mm_mul_ps(vec_previous, vec_w_previous);
vec_out = _mm_add_ps(vec_out, _mm_mul_ps(vec_current, vec_w_current));
vec_out =
_mm_add_ps(vec_out, _mm_shuffle_ps(vec_out,vec_out,_MM_SHUFFLE(3, 2, 3, 2)));
_mm_storel_pi((__m64*)outp,vec_out);
re->distance += ratio;
outp++;
}
re->distance -= 1.0;
inp++;
}
_mm_storeu_ps((float*)&re->buffer[0], vec_previous);
_mm_storeu_ps((float*)&re->buffer[2], vec_current);
data->output_frames = outp - (audio_frame_float_t*)data->data_out;
}
#elif defined (__ARM_NEON__)
#define CC_RESAMPLER_IDENT "NEON"
size_t resampler_CC_downsample_neon(float *outp, const float *inp,
rarch_CC_resampler_t* re_, size_t input_frames, float ratio);
size_t resampler_CC_upsample_neon (float *outp, const float *inp,
rarch_CC_resampler_t* re_, size_t input_frames, float ratio);
static void resampler_CC_downsample(void *re_, struct resampler_data *data)
{
data->output_frames = resampler_CC_downsample_neon(
data->data_out, data->data_in, re_, data->input_frames, data->ratio);
}
static void resampler_CC_upsample(void *re_, struct resampler_data *data)
{
data->output_frames = resampler_CC_upsample_neon(
data->data_out, data->data_in, re_, data->input_frames, data->ratio);
}
#else
/* C reference version. Not optimized. */
#define CC_RESAMPLER_IDENT "C"
#if (CC_RESAMPLER_PRECISION > 4)
static INLINE float cc_int(float x, float b)
{
float val = x * b * M_PI + sinf(x * b * M_PI);
return (val > M_PI) ? M_PI : (val < -M_PI) ? -M_PI : val;
}
#define cc_kernel(x, b) ((cc_int((x) + 0.5, (b)) - cc_int((x) - 0.5, (b))) / (2.0 * M_PI))
#else
static INLINE float cc_int(float x, float b)
{
float val = x * b;
#if (CC_RESAMPLER_PRECISION > 0)
val = val*(1 - 0.25 * val * val * (3.0 - val * val));
#endif
return (val > 0.5) ? 0.5 : (val < -0.5) ? -0.5 : val;
}
#define cc_kernel(x, b) ((cc_int((x) + 0.5, (b)) - cc_int((x) - 0.5, (b))))
#endif
static INLINE void add_to(const audio_frame_float_t *source,
audio_frame_float_t *target, float ratio)
{
target->l += source->l * ratio;
target->r += source->r * ratio;
}
static void resampler_CC_downsample(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = (audio_frame_float_t*)
(inp + data->input_frames);
audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
float ratio = 1.0 / data->ratio;
float b = data->ratio; /* cutoff frequency. */
while (inp != inp_max)
{
add_to(inp, re->buffer + 0, cc_kernel(re->distance, b));
add_to(inp, re->buffer + 1, cc_kernel(re->distance - ratio, b));
add_to(inp, re->buffer + 2, cc_kernel(re->distance - ratio - ratio, b));
re->distance++;
inp++;
if (re->distance > (ratio + 0.5))
{
*outp = re->buffer[0];
re->buffer[0] = re->buffer[1];
re->buffer[1] = re->buffer[2];
re->buffer[2].l = 0.0;
re->buffer[2].r = 0.0;
re->distance -= ratio;
outp++;
}
}
data->output_frames = outp - (audio_frame_float_t*)data->data_out;
}
static void resampler_CC_upsample(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = (audio_frame_float_t*)
(inp + data->input_frames);
audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
float b = min(data->ratio, 1.00); /* cutoff frequency. */
float ratio = 1.0 / data->ratio;
while (inp != inp_max)
{
re->buffer[0] = re->buffer[1];
re->buffer[1] = re->buffer[2];
re->buffer[2] = re->buffer[3];
re->buffer[3] = *inp;
while (re->distance < 1.0)
{
int i;
outp->l = 0.0;
outp->r = 0.0;
for (i = 0; i < 4; i++)
{
float temp = cc_kernel(re->distance + 1.0 - i, b);
outp->l += re->buffer[i].l * temp;
outp->r += re->buffer[i].r * temp;
}
re->distance += ratio;
outp++;
}
re->distance -= 1.0;
inp++;
}
data->output_frames = outp - (audio_frame_float_t*)data->data_out;
}
#endif
static void resampler_CC_process(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
if (re)
re->process(re_, data);
}
static void *resampler_CC_init(const struct resampler_config *config,
double bandwidth_mod, resampler_simd_mask_t mask)
{
int i;
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)
memalign_alloc(32, sizeof(rarch_CC_resampler_t));
/* TODO: lookup if NEON support can be detected at
* runtime and a funcptr set at runtime for either
* C codepath or NEON codepath. This will help out
* Android. */
(void)mask;
(void)config;
if (!re)
return NULL;
for (i = 0; i < 4; i++)
{
re->buffer[i].l = 0.0;
re->buffer[i].r = 0.0;
}
/* Variations of data->ratio around 0.75 are safer
* than around 1.0 for both up/downsampler. */
if (bandwidth_mod < 0.75)
{
re->process = resampler_CC_downsample;
re->distance = 0.0;
}
else
{
re->process = resampler_CC_upsample;
re->distance = 2.0;
}
return re;
}
#endif
static void resampler_CC_free(void *re_)
{
#ifndef _MIPS_ARCH_ALLEGREX
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
if (re)
memalign_free(re);
#endif
(void)re_;
}
rarch_resampler_t CC_resampler = {
resampler_CC_init,
resampler_CC_process,
resampler_CC_free,
RESAMPLER_API_VERSION,
"CC",
"cc"
};
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