mirror of
https://github.com/libretro/RetroArch.git
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ba7cefc529
fix overflow in VFPU resampler when input_frames is 0 add support for GU_PSM_5551 pixel format when using hardware rendering
323 lines
8.6 KiB
C
323 lines
8.6 KiB
C
/* 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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* 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 <math.h>
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#include <stdint.h>
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#include <stdlib.h>
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#include <stdio.h>
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#if !defined(RESAMPLER_TEST) && defined(RARCH_INTERNAL)
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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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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_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 = (audio_frame_float_t*)(inp + data->input_frames);
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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__asm__ (
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".set push\n"
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".set noreorder\n"
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"mtv %2, s700 \n" // 700 = data->ratio = b
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// "vsat0.s s700, s700 \n"
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"vrcp.s s701, s700 \n" // 701 = 1.0 / b
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"vadd.s s702, s700, s700 \n" // 702 = 2 * b
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"vmul.s s703, s700, s710 \n" // 703 = b * pi
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"mfv %0, s701 \n"
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"mfv %1, s730 \n"
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".set pop\n"
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: "=r"(ratio), "=r"(fraction)
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: "r"((float)data->ratio)
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);
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for (;;)
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{
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while (fraction < ratio)
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{
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if (inp == inp_max)
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goto done;
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__asm__ (
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".set push \n"
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".set noreorder \n"
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"lv.s s620, 0(%1) \n"
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"lv.s s621, 4(%1) \n"
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"vsub.s s731, s701, s730 \n"
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"vadd.q c600, c730[-X,Y,-X,Y], c730[1/2,1/2,-1/2,-1/2]\n"
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"vmul.q c610, c600, c700[Z,Z,Z,Z] \n" //*2*b
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"vmul.q c600, c600, c700[W,W,W,W] \n" //*b*pi
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"vsin.q c610, c610 \n"
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"vadd.q c600, c600, c610 \n"
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"vmul.q c600[-1:1,-1:1,-1:1,-1:1], c600, c710[Y,Y,Y,Y] \n"
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"vsub.p c600, c600, c602 \n"
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"vmul.q c620, c620[X,Y,X,Y], c600[X,X,Y,Y] \n"
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"vadd.q c720, c720, c620 \n"
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"vadd.s s730, s730, s730[1] \n"
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"mfv %0, s730 \n"
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".set pop \n"
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: "=r"(fraction)
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: "r"(inp));
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inp++;
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}
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__asm__ (
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".set push \n"
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".set noreorder \n"
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"vmul.p c720, c720, c720[1/2,1/2] \n"
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"sv.s s720, 0(%1) \n"
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"sv.s s721, 4(%1) \n"
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"vmov.q c720, c720[Z,W,0,0] \n"
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"vsub.s s730, s730, s701 \n"
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"mfv %0, s730 \n"
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".set pop \n"
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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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}
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static void resampler_CC_free(void *re_)
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{
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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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__asm__ (
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".set push\n"
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".set noreorder\n"
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"vcst.s s710, VFPU_PI \n" // 710 = pi
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"vcst.s s711, VFPU_1_PI \n" // 711 = 1.0 / (pi)
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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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RARCH_LOG("\nConvoluted Cosine resampler (VFPU): \n");
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return (void*)-1;
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}
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#else
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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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{
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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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{
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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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{
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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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float ratio, b;
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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 = (audio_frame_float_t*)(inp + data->input_frames);
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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ratio = 1.0 / data->ratio;
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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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float b, ratio;
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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 = (audio_frame_float_t*)(inp + data->input_frames);
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audio_frame_float_t *outp = (audio_frame_float_t*)data->data_out;
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b = min(data->ratio, 1.00); // cutoff frequency
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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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resampler_CC_process,
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resampler_CC_free,
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"CC",
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};
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