mirror/RetroArch
1
0
Fork 0
mirror of https://github.com/libretro/RetroArch synced 2025-03-02 19:13:34 +00:00
Code Issues Packages Projects Releases Wiki Activity

(PSP) add a C reference implementation of the VFPU audio resampler.

Browse Source 
asm version still lacks some improvements found in the C version.
This commit is contained in:
aliaspider 2014-03-19 20:52:15 +01:00
parent e060b67136
commit dc95cf4c82
1 changed files with 213 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

235
audio/cc_resampler.c
View File

@ -13,35 +13,37 @@
#include <stdlib.h>
#include <stdio.h>
#ifndef RESAMPLER_TEST
#include "../general.h"
#else
#define RARCH_LOG(...) fprintf(stderr, __VA_ARGS__)
#endif
#ifdef _MIPS_ARCH_ALLEGREX1
typedef struct rarch_CC_resampler
{
int dummy;
}rarch_CC_resampler_t;
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_ALLEGREX
static void resampler_CC_process(void *re_, struct resampler_data *data)
{
(void)re_;
// rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
float ratio,fraction;
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;
audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
audio_frame_float_t *inp_max = inp + data->input_frames;
@ -123,9 +125,7 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
done:
data->output_frames = (outp - (audio_frame_float_t*)data->data_out);
}
#else
#error "platform not supported"
#endif
static void resampler_CC_free(void *re_)
{
@ -153,8 +153,202 @@ static void *resampler_CC_init(double bandwidth_mod)
".set pop\n"
);
RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
return re;
}
#else
//#define HAVE_SSE_MATHFUN_H
#if defined(__SSE2__) && defined(HAVE_SSE_MATHFUN_H)
#define USE_SSE2
#include "sse_mathfun.h"
static inline float _mm_sin(float x)
{
static float temp;
__m128 vector = _mm_set1_ps(x);
vector = sin_ps(vector);
_mm_store1_ps(&temp,vector);
return temp;
}
static inline float _mm_cos(float x)
{
static float temp;
__m128 vector = _mm_set1_ps(x);
vector = cos_ps(vector);
_mm_store1_ps(&temp,vector);
return temp;
}
#define sin(x) _mm_sin(x)
#define cos(x) _mm_cos(x)
#endif
typedef struct audio_frame_float
{
float l;
float r;
}audio_frame_float_t;
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 + sin(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)
{
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 = 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);
}
#ifndef min
#define min(a, b) ((a) < (b) ? (a) : (b))
#endif
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 = 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;
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,
@ -162,6 +356,3 @@ const rarch_resampler_t CC_resampler = {
resampler_CC_free,
"CC",
};