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https://github.com/libretro/RetroArch
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(PSP) add a C reference implementation of the VFPU audio resampler.
asm version still lacks some improvements found in the C version.
This commit is contained in:
aliaspider
parent
e060b67136
commit
dc95cf4c82
@ -13,35 +13,37 @@
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#include <stdlib.h>
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#include <stdio.h>
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#ifndef RESAMPLER_TEST
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#include "../general.h"
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#else
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#define RARCH_LOG(...) fprintf(stderr, __VA_ARGS__)
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#endif
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#ifdef _MIPS_ARCH_ALLEGREX1
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typedef struct rarch_CC_resampler
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{
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int dummy;
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}rarch_CC_resampler_t;
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typedef struct audio_frame_float
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{
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float l;
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float r;
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}audio_frame_float_t;
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typedef struct audio_frame_int16
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{
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int16_t l;
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int16_t r;
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}audio_frame_int16_t;
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#ifdef _MIPS_ARCH_ALLEGREX
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static void resampler_CC_process(void *re_, struct resampler_data *data)
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{
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(void)re_;
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// rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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float ratio,fraction;
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typedef struct audio_frame_float
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{
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float l;
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float r;
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}audio_frame_float_t;
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typedef struct audio_frame_int16
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{
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int16_t l;
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int16_t r;
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}audio_frame_int16_t;
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audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
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audio_frame_float_t *inp_max = inp + data->input_frames;
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@ -123,9 +125,7 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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done:
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data->output_frames = (outp - (audio_frame_float_t*)data->data_out);
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}
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#else
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#error "platform not supported"
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#endif
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static void resampler_CC_free(void *re_)
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{
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@ -153,8 +153,202 @@ static void *resampler_CC_init(double bandwidth_mod)
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".set pop\n"
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);
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RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
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return re;
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}
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#else
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//#define HAVE_SSE_MATHFUN_H
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#if defined(__SSE2__) && defined(HAVE_SSE_MATHFUN_H)
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#define USE_SSE2
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#include "sse_mathfun.h"
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static inline float _mm_sin(float x)
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{
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static float temp;
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__m128 vector = _mm_set1_ps(x);
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vector = sin_ps(vector);
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_mm_store1_ps(&temp,vector);
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return temp;
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}
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static inline float _mm_cos(float x)
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{
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static float temp;
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__m128 vector = _mm_set1_ps(x);
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vector = cos_ps(vector);
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_mm_store1_ps(&temp,vector);
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return temp;
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}
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#define sin(x) _mm_sin(x)
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#define cos(x) _mm_cos(x)
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#endif
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typedef struct audio_frame_float
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{
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float l;
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float r;
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}audio_frame_float_t;
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typedef struct rarch_CC_resampler
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{
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audio_frame_float_t buffer[4];
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float distance;
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void (*process)(void *re, struct resampler_data *data);
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}rarch_CC_resampler_t;
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static inline float cc_int(float x, float b){
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float val = x * b * M_PI + sin(x * b * M_PI);
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return (val > M_PI)? M_PI : (val < -M_PI)? -M_PI : val;
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}
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static inline float cc_kernel(float x, float b){
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return (cc_int(x + 0.5, b) - cc_int(x - 0.5, b)) / (2.0 * M_PI);
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}
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static inline void add_to(const audio_frame_float_t* source,audio_frame_float_t* target, float ratio){
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target->l += source->l * ratio;
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target->r += source->r * ratio;
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}
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static void resampler_CC_downsample(void *re_, struct resampler_data *data)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
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audio_frame_float_t *inp_max = inp + data->input_frames;
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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float ratio = 1.0 / data->ratio;
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float b = data->ratio; // cutoff frequency
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while(inp != inp_max)
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{
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add_to(inp, re->buffer + 0, cc_kernel(re->distance, b));
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add_to(inp, re->buffer + 1, cc_kernel(re->distance - ratio, b));
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add_to(inp, re->buffer + 2, cc_kernel(re->distance - ratio - ratio, b));
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re->distance++;
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inp++;
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if (re->distance > (ratio + 0.5))
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{
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*outp=re->buffer[0];
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re->buffer[0] = re->buffer[1];
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re->buffer[1] = re->buffer[2];
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re->buffer[2].l = 0.0;
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re->buffer[2].r = 0.0;
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re->distance-=ratio;
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outp++;
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}
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}
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data->output_frames = (outp - (audio_frame_float_t*)data->data_out);
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}
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#ifndef min
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#define min(a, b) ((a) < (b) ? (a) : (b))
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#endif
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static void resampler_CC_upsample(void *re_, struct resampler_data *data)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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audio_frame_float_t *inp = (audio_frame_float_t*)data->data_in;
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audio_frame_float_t *inp_max = inp + data->input_frames;
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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float b = min(data->ratio, 1.00); // cutoff frequency
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float ratio = 1.0 / data->ratio;
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while(inp != inp_max)
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{
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re->buffer[0] = re->buffer[1];
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re->buffer[1] = re->buffer[2];
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re->buffer[2] = re->buffer[3];
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re->buffer[3] = *inp;
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while (re->distance < 1.0)
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{
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int i;
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float temp;
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outp->l = 0.0;
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outp->r = 0.0;
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for (i=0; i!=4; i++)
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{
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temp = cc_kernel(re->distance + 1.0 - i, b);
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outp->l += re->buffer[i].l * temp;
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outp->r += re->buffer[i].r * temp;
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}
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re->distance += ratio;
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outp++;
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}
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re->distance-= 1.0;
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inp++;
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}
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data->output_frames = (outp - (audio_frame_float_t*)data->data_out);
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}
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static void resampler_CC_process(void *re_, struct resampler_data *data)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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re->process(re_, data);
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}
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static void resampler_CC_free(void *re_)
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{
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_;
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if (re)
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free(re);
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}
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static void *resampler_CC_init(double bandwidth_mod)
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{
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int i;
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rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)calloc(1, sizeof(rarch_CC_resampler_t));
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if (!re)
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return NULL;
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for (i=0; i!=4 ; i++)
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{
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re->buffer[i].l=0.0;
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re->buffer[i].r=0.0;
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}
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RARCH_LOG("Convoluted Cosine resampler (C) : ");
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if (bandwidth_mod < 0.75) // variations of data->ratio around 0.75 are safer than around 1.0 for both up/downsampler.
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{
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RARCH_LOG("CC_downsample @%f \n", bandwidth_mod);
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re->process = resampler_CC_downsample;
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re->distance = 0.0;
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}
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else
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{
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RARCH_LOG("CC_upsample @%f \n", bandwidth_mod);
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re->process = resampler_CC_upsample;
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re->distance = 2.0;
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}
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return re;
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}
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#endif
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const rarch_resampler_t CC_resampler = {
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resampler_CC_init,
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@ -162,6 +356,3 @@ const rarch_resampler_t CC_resampler = {
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resampler_CC_free,
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"CC",
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};
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