/* RetroArch - A frontend for libretro. * Copyright (C) 2014-2016 - 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 . */ /* Convoluted Cosine Resampler */ #include #include #ifdef __SSE__ #include #endif #include #include #include #include "../audio_resampler_driver.h" /* Since SSE and NEON don't provide support for trigonometric functions * we approximate those with polynoms * * CC_RESAMPLER_PRECISION defines how accurate the approximation is * a setting of 5 or more means full precison. * setting 0 doesn't use a polynom * setting 1 uses P(X) = X - (3/4)*X^3 + (1/4)*X^5 * * only 0 and 1 are implemented for SSE and NEON currently * * the MIPS_ARCH_ALLEGREX target doesnt require this setting since it has * native support for the required functions so it will always use full precision. */ #ifndef CC_RESAMPLER_PRECISION #define CC_RESAMPLER_PRECISION 1 #endif 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; #ifdef _MIPS_ARCH_ALLEGREX static void resampler_CC_process(void *re_, struct resampler_data *data) { 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; (void)re_; __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_init(const struct resampler_config *config, double bandwidth_mod, resampler_simd_mask_t mask) { (void)mask; (void)bandwidth_mod; (void)config; __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"); return (void*)-1; } #else #if defined(__SSE__) #define CC_RESAMPLER_IDENT "SSE" 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 = (audio_frame_float_t*)(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. */ __m128 vec_previous = _mm_loadu_ps((float*)&re->buffer[0]); __m128 vec_current = _mm_loadu_ps((float*)&re->buffer[2]); while (inp != inp_max) { __m128 vec_ww1, vec_ww2; __m128 vec_w_previous; __m128 vec_w_current; __m128 vec_in; __m128 vec_ratio = _mm_mul_ps(_mm_set_ps1(ratio), _mm_set_ps(3.0, 2.0, 1.0, 0.0)); __m128 vec_w = _mm_sub_ps(_mm_set_ps1(re->distance), vec_ratio); __m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5)); __m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5)); __m128 vec_b = _mm_set_ps1(b); vec_w1 = _mm_mul_ps(vec_w1, vec_b); vec_w2 = _mm_mul_ps(vec_w2, vec_b); (void)vec_ww1; (void)vec_ww2; #if (CC_RESAMPLER_PRECISION > 0) vec_ww1 = _mm_mul_ps(vec_w1, vec_w1); vec_ww2 = _mm_mul_ps(vec_w2, vec_w2); vec_ww1 = _mm_mul_ps(vec_ww1, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww1)); vec_ww2 = _mm_mul_ps(vec_ww2, _mm_sub_ps(_mm_set_ps1(3.0),vec_ww2)); vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww1); vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0/4.0), vec_ww2); vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1)); vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2)); #endif vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5)); vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5)); vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5)); vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5)); vec_w = _mm_sub_ps(vec_w1, vec_w2); vec_w_previous = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0)); vec_w_current = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2)); vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp); vec_in = _mm_shuffle_ps(vec_in,vec_in,_MM_SHUFFLE(1, 0, 1, 0)); vec_previous = _mm_add_ps(vec_previous, _mm_mul_ps(vec_in, vec_w_previous)); vec_current = _mm_add_ps(vec_current, _mm_mul_ps(vec_in, vec_w_current)); re->distance++; inp++; if (re->distance > (ratio + 0.5)) { _mm_storel_pi((__m64*)outp, vec_previous); vec_previous = _mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2)); vec_current = _mm_shuffle_ps(vec_current,_mm_setzero_ps(),_MM_SHUFFLE(1, 0, 3, 2)); re->distance -= ratio; outp++; } } _mm_storeu_ps((float*)&re->buffer[0], vec_previous); _mm_storeu_ps((float*)&re->buffer[2], vec_current); data->output_frames = outp - (audio_frame_float_t*)data->data_out; } 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 = (audio_frame_float_t*)(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; __m128 vec_previous = _mm_loadu_ps((float*)&re->buffer[0]); __m128 vec_current = _mm_loadu_ps((float*)&re->buffer[2]); while (inp != inp_max) { __m128 vec_in = _mm_loadl_pi(_mm_setzero_ps(),(__m64*)inp); vec_previous = _mm_shuffle_ps(vec_previous,vec_current,_MM_SHUFFLE(1, 0, 3, 2)); vec_current = _mm_shuffle_ps(vec_current,vec_in,_MM_SHUFFLE(1, 0, 3, 2)); while (re->distance < 1.0) { __m128 vec_w_previous, vec_w_current, vec_out; #if (CC_RESAMPLER_PRECISION > 0) __m128 vec_ww1, vec_ww2; #endif __m128 vec_w = _mm_add_ps(_mm_set_ps1(re->distance), _mm_set_ps(-2.0, -1.0, 0.0, 1.0)); __m128 vec_w1 = _mm_add_ps(vec_w , _mm_set_ps1(0.5)); __m128 vec_w2 = _mm_sub_ps(vec_w , _mm_set_ps1(0.5)); __m128 vec_b = _mm_set_ps1(b); vec_w1 = _mm_mul_ps(vec_w1, vec_b); vec_w2 = _mm_mul_ps(vec_w2, vec_b); #if (CC_RESAMPLER_PRECISION > 0) vec_ww1 = _mm_mul_ps(vec_w1, vec_w1); vec_ww2 = _mm_mul_ps(vec_w2, vec_w2); vec_ww1 = _mm_mul_ps(vec_ww1,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww1)); vec_ww2 = _mm_mul_ps(vec_ww2,_mm_sub_ps(_mm_set_ps1(3.0),vec_ww2)); vec_ww1 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww1); vec_ww2 = _mm_mul_ps(_mm_set_ps1(1.0 / 4.0), vec_ww2); vec_w1 = _mm_mul_ps(vec_w1, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww1)); vec_w2 = _mm_mul_ps(vec_w2, _mm_sub_ps(_mm_set_ps1(1.0), vec_ww2)); #endif vec_w1 = _mm_min_ps(vec_w1, _mm_set_ps1( 0.5)); vec_w2 = _mm_min_ps(vec_w2, _mm_set_ps1( 0.5)); vec_w1 = _mm_max_ps(vec_w1, _mm_set_ps1(-0.5)); vec_w2 = _mm_max_ps(vec_w2, _mm_set_ps1(-0.5)); vec_w = _mm_sub_ps(vec_w1, vec_w2); vec_w_previous = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(1, 1, 0, 0)); vec_w_current = _mm_shuffle_ps(vec_w,vec_w,_MM_SHUFFLE(3, 3, 2, 2)); vec_out = _mm_mul_ps(vec_previous, vec_w_previous); vec_out = _mm_add_ps(vec_out, _mm_mul_ps(vec_current, vec_w_current)); vec_out = _mm_add_ps(vec_out, _mm_shuffle_ps(vec_out,vec_out,_MM_SHUFFLE(3, 2, 3, 2))); _mm_storel_pi((__m64*)outp,vec_out); re->distance += ratio; outp++; } re->distance -= 1.0; inp++; } _mm_storeu_ps((float*)&re->buffer[0], vec_previous); _mm_storeu_ps((float*)&re->buffer[2], vec_current); data->output_frames = outp - (audio_frame_float_t*)data->data_out; } #elif defined (__ARM_NEON__) #define CC_RESAMPLER_IDENT "NEON" size_t resampler_CC_downsample_neon(float *outp, const float *inp, rarch_CC_resampler_t* re_, size_t input_frames, float ratio); size_t resampler_CC_upsample_neon (float *outp, const float *inp, rarch_CC_resampler_t* re_, size_t input_frames, float ratio); static void resampler_CC_downsample(void *re_, struct resampler_data *data) { data->output_frames = resampler_CC_downsample_neon( data->data_out, data->data_in, re_, data->input_frames, data->ratio); } static void resampler_CC_upsample(void *re_, struct resampler_data *data) { data->output_frames = resampler_CC_upsample_neon( data->data_out, data->data_in, re_, data->input_frames, data->ratio); } #else /* C reference version. Not optimized. */ #define CC_RESAMPLER_IDENT "C" #if (CC_RESAMPLER_PRECISION > 4) 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; } #define cc_kernel(x, b) ((cc_int((x) + 0.5, (b)) - cc_int((x) - 0.5, (b))) / (2.0 * M_PI)) #else static INLINE float cc_int(float x, float b) { float val = x * b; #if (CC_RESAMPLER_PRECISION > 0) val = val*(1 - 0.25 * val * val * (3.0 - val * val)); #endif return (val > 0.5) ? 0.5 : (val < -0.5) ? -0.5 : val; } #define cc_kernel(x, b) ((cc_int((x) + 0.5, (b)) - cc_int((x) - 0.5, (b)))) #endif 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 = (audio_frame_float_t*) (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; } 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 = (audio_frame_float_t*) (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; outp->l = 0.0; outp->r = 0.0; for (i = 0; i < 4; i++) { float 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; } #endif static void resampler_CC_process(void *re_, struct resampler_data *data) { rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_; if (re) re->process(re_, data); } static void *resampler_CC_init(const struct resampler_config *config, double bandwidth_mod, resampler_simd_mask_t mask) { int i; rarch_CC_resampler_t *re = (rarch_CC_resampler_t*) memalign_alloc(32, sizeof(rarch_CC_resampler_t)); /* TODO: lookup if NEON support can be detected at * runtime and a funcptr set at runtime for either * C codepath or NEON codepath. This will help out * Android. */ (void)mask; (void)config; if (!re) return NULL; for (i = 0; i < 4; i++) { re->buffer[i].l = 0.0; re->buffer[i].r = 0.0; } /* Variations of data->ratio around 0.75 are safer * than around 1.0 for both up/downsampler. */ if (bandwidth_mod < 0.75) { re->process = resampler_CC_downsample; re->distance = 0.0; } else { re->process = resampler_CC_upsample; re->distance = 2.0; } return re; } #endif static void resampler_CC_free(void *re_) { #ifndef _MIPS_ARCH_ALLEGREX rarch_CC_resampler_t *re = (rarch_CC_resampler_t*)re_; if (re) memalign_free(re); #endif (void)re_; } rarch_resampler_t CC_resampler = { resampler_CC_init, resampler_CC_process, resampler_CC_free, RESAMPLER_API_VERSION, "CC", "cc" };