RetroArch/audio/cc_resampler.c
aliaspider ba7cefc529 (PSP) add BIG_STACK makefile option
fix overflow in VFPU resampler when input_frames is 0
add support for GU_PSM_5551 pixel format when using hardware rendering
2014-06-29 05:45:36 +01:00

323 lines
8.6 KiB
C

/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
* Copyright (C) 2014 - 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 "resampler.h"
#include <math.h>
#include <stdint.h>
#include <stdlib.h>
#include <stdio.h>
#if !defined(RESAMPLER_TEST) && defined(RARCH_INTERNAL)
#include "../general.h"
#else
#define RARCH_LOG(...) fprintf(stderr, __VA_ARGS__)
#endif
typedef struct audio_frame_float
{
float l;
float r;
} audio_frame_float_t;
typedef struct audio_frame_int16
{
int16_t l;
int16_t r;
} audio_frame_int16_t;
#ifdef _MIPS_ARCH_ALLEGREX1
static void resampler_CC_process(void *re_, struct resampler_data *data)
{
(void)re_;
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;
__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_free(void *re_)
{
(void)re_;
}
static void *resampler_CC_init(double bandwidth_mod)
{
__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");
RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
return (void*)-1;
}
#else
// C reference version. Not optimized.
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;
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;
}
static inline float cc_kernel(float x, float b)
{
return (cc_int(x + 0.5, b) - cc_int(x - 0.5, b)) / (2.0 * M_PI);
}
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)
{
float ratio, b;
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;
ratio = 1.0 / data->ratio;
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;
}
#ifndef min
#define min(a, b) ((a) < (b) ? (a) : (b))
#endif
static void resampler_CC_upsample(void *re_, struct resampler_data *data)
{
float b, ratio;
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;
b = min(data->ratio, 1.00); // cutoff frequency
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;
float temp;
outp->l = 0.0;
outp->r = 0.0;
for (i = 0; i < 4; i++)
{
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;
}
static void resampler_CC_process(void *re_, struct resampler_data *data)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
re->process(re_, data);
}
static void resampler_CC_free(void *re_)
{
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
if (re)
free(re);
}
static void *resampler_CC_init(double bandwidth_mod)
{
int i;
rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)calloc(1, sizeof(rarch_CC_resampler_t));
if (!re)
return NULL;
for (i = 0; i < 4; i++)
{
re->buffer[i].l = 0.0;
re->buffer[i].r = 0.0;
}
RARCH_LOG("Convoluted Cosine resampler (C) : ");
if (bandwidth_mod < 0.75) // variations of data->ratio around 0.75 are safer than around 1.0 for both up/downsampler.
{
RARCH_LOG("CC_downsample @%f \n", bandwidth_mod);
re->process = resampler_CC_downsample;
re->distance = 0.0;
}
else
{
RARCH_LOG("CC_upsample @%f \n", bandwidth_mod);
re->process = resampler_CC_upsample;
re->distance = 2.0;
}
return re;
}
#endif
const rarch_resampler_t CC_resampler = {
resampler_CC_init,
resampler_CC_process,
resampler_CC_free,
"CC",
};