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https://github.com/libretro/RetroArch
synced 2025-02-21 18:40:09 +00:00
Stylistic cleanups in CC resampler.
This commit is contained in:
Themaister
parent
0c57a1726b
commit
4d9ff7d147
@ -1,10 +1,21 @@
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/*
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* Convoluted Cosine Resampler
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/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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* Copyright (C) 2014 - Ali Bouhlel ( aliaspider@gmail.com )
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*
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* licence: GPLv3
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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// Convoluted Cosine Resampler
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#include "resampler.h"
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#include "../libretro.h"
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#include "../performance.h"
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@ -19,19 +30,6 @@
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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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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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@ -44,6 +42,11 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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int16_t r;
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} audio_frame_int16_t;
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#ifdef _MIPS_ARCH_ALLEGREX1
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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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float ratio, fraction;
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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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@ -63,12 +66,13 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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"mfv %1, s730 \n"
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".set pop\n"
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:"=r"(ratio),"=r"(fraction): "r"((float)data->ratio)
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: "=r"(ratio), "=r"(fraction)
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: "r"((float)data->ratio)
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);
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while(true)
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for (;;)
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{
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while ((fraction < ratio))
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while (fraction < ratio)
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{
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__asm__ (
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".set push \n"
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@ -99,8 +103,9 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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"mfv %0, s730 \n"
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".set pop \n"
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:"=r"(fraction): "r"(inp)
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);
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: "=r"(fraction)
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: "r"(inp));
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inp++;
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if (inp == inp_max)
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goto done;
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@ -117,29 +122,26 @@ static void resampler_CC_process(void *re_, struct resampler_data *data)
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"mfv %0, s730 \n"
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".set pop \n"
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:"=r"(fraction): "r"(outp)
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);
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: "=r"(fraction)
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: "r"(outp));
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outp++;
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}
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// The VFPU state is assumed to remain intact in-between calls to resampler_CC_process.
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done:
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data->output_frames = (outp - (audio_frame_float_t*)data->data_out);
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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_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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(void)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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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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__asm__ (
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".set push\n"
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".set noreorder\n"
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@ -150,75 +152,40 @@ static void *resampler_CC_init(double bandwidth_mod)
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"vzero.q c720 \n"
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"vzero.q c730 \n"
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".set pop\n"
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);
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".set pop\n");
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RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
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return re;
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return (void*)-1;
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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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// C reference version. Not optimized.
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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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static inline float cc_int(float x, float b)
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{
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float val = x * b * M_PI + sinf(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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static inline float cc_kernel(float x, float b)
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{
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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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static inline void add_to(const audio_frame_float_t *source, audio_frame_float_t *target, float ratio)
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{
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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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@ -253,7 +220,7 @@ static void resampler_CC_downsample(void *re_, struct resampler_data *data)
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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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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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@ -262,7 +229,6 @@ static void resampler_CC_downsample(void *re_, struct resampler_data *data)
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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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@ -286,7 +252,7 @@ static void resampler_CC_upsample(void *re_, struct resampler_data *data)
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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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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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@ -301,8 +267,7 @@ static void resampler_CC_upsample(void *re_, struct resampler_data *data)
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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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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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@ -325,7 +290,7 @@ static void *resampler_CC_init(double bandwidth_mod)
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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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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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@ -356,3 +321,4 @@ 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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@ -1,5 +1,6 @@
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/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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* Copyright (C) 2014 - Ali Bouhlel ( aliaspider@gmail.com )
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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@ -167,11 +168,12 @@ static void audio_convert_float_to_s16_neon(int16_t *out, const float *in, size_
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void audio_convert_s16_to_float_ALLEGREX(float *out,
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const int16_t *in, size_t samples, float gain)
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{
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#ifdef DEBUG
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// make sure the buffer is 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK
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rarch_assert(((uint32_t)out & 0xF) == 0);
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// Make sure the buffer is 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK.
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// Only the output buffer can be assumed to be 16-byte aligned.
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rarch_assert(((uintptr_t)out & 0xf) == 0);
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#endif
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size_t i;
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gain = gain / 0x8000;
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__asm__ (
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@ -179,10 +181,9 @@ void audio_convert_s16_to_float_ALLEGREX(float *out,
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".set noreorder \n"
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"mtv %0, s200 \n"
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".set pop \n"
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::"r"(gain)
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);
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::"r"(gain));
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for (i = 0; (i+16) <= samples; i+=16)
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for (i = 0; i + 16 <= samples; i += 16)
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{
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__asm__ (
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".set push \n"
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@ -215,11 +216,10 @@ void audio_convert_s16_to_float_ALLEGREX(float *out,
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"sv.q c130, 48(%1) \n"
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".set pop \n"
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::"r"(in+i),"r"(out+i)
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);
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:: "r"(in + i), "r"(out + i));
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}
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for (;i != samples; i++)
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for (; i < samples; i++)
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out[i] = (float)in[i] * gain;
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}
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@ -227,13 +227,14 @@ void audio_convert_float_to_s16_ALLEGREX(int16_t *out,
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const float *in, size_t samples)
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{
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#ifdef DEBUG
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// make sure the buffers are 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK
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rarch_assert(((uint32_t)in & 0xF) == 0);
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rarch_assert(((uint32_t)out & 0xF) == 0);
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// Make sure the buffers are 16 byte aligned, this should be the default behaviour of malloc in the PSPSDK.
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// Both buffers are allocated by RetroArch, so can assume alignment.
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rarch_assert(((uintptr_t)in & 0xf) == 0);
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rarch_assert(((uintptr_t)out & 0xf) == 0);
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#endif
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size_t i;
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for (i = 0; (i+8) <= samples; i+=8)
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for (i = 0; i + 8 <= samples; i += 8)
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{
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__asm__ (
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".set push \n"
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@ -250,16 +251,14 @@ void audio_convert_float_to_s16_ALLEGREX(int16_t *out,
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"sv.q c100, 0(%1) \n"
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".set pop \n"
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::"r"(in+i),"r"(out+i)
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);
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:: "r"(in + i), "r"(out + i));
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}
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for (;i != samples; i++)
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for (; i < samples; i++)
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{
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int32_t val = (int32_t)(in[i] * 0x8000);
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out[i] = (val > 0x7FFF) ? 0x7FFF : (val < -0x8000 ? -0x8000 : (int16_t)val);
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}
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}
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#endif
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