From b4984765317d8d2a27e4b58eda946a0b3f15af8e Mon Sep 17 00:00:00 2001 From: Themaister Date: Tue, 20 May 2014 12:34:50 +0200 Subject: [PATCH] Remove broken DSP plugs. They will all have to be properly rewritten for new interface. --- audio/dsp_filter.c | 2 +- audio/filters/boolean.h | 33 ---- audio/filters/echo.c | 187 ------------------ audio/filters/eq.c | 422 ---------------------------------------- audio/filters/iir.c | 421 --------------------------------------- audio/filters/phaser.c | 193 ------------------ audio/filters/reverb.c | 397 ------------------------------------- audio/filters/volume.c | 131 ------------- audio/filters/wah.c | 187 ------------------ driver.h | 10 +- griffin/griffin.c | 13 +- 11 files changed, 4 insertions(+), 1992 deletions(-) delete mode 100644 audio/filters/boolean.h delete mode 100644 audio/filters/echo.c delete mode 100644 audio/filters/eq.c delete mode 100644 audio/filters/iir.c delete mode 100644 audio/filters/phaser.c delete mode 100644 audio/filters/reverb.c delete mode 100644 audio/filters/volume.c delete mode 100644 audio/filters/wah.c diff --git a/audio/dsp_filter.c b/audio/dsp_filter.c index d1693a75d1..8d7b2c5fd6 100644 --- a/audio/dsp_filter.c +++ b/audio/dsp_filter.c @@ -225,7 +225,7 @@ static bool append_plugs(rarch_dsp_filter_t *dsp, struct string_list *list) unsigned i; dspfilter_simd_mask_t mask = rarch_get_cpu_features(); - for (unsigned i = 0; i < dsp->num_plugs; i++) + for (i = 0; i < dsp->num_plugs; i++) { dylib_t lib = dylib_load(list->elems[i].data); if (!lib) diff --git a/audio/filters/boolean.h b/audio/filters/boolean.h deleted file mode 100644 index b0a20bd8d6..0000000000 --- a/audio/filters/boolean.h +++ /dev/null @@ -1,33 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * - * 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 . - */ - -#ifndef __RARCH_BOOLEAN_H -#define __RARCH_BOOLEAN_H - -#ifndef __cplusplus - -#if defined(_MSC_VER) && !defined(SN_TARGET_PS3) -/* Hack applied for MSVC when compiling in C89 mode as it isn't C99 compliant. */ -#define bool unsigned char -#define true 1 -#define false 0 -#else -#include -#endif - -#endif - -#endif - diff --git a/audio/filters/echo.c b/audio/filters/echo.c deleted file mode 100644 index b809c9505c..0000000000 --- a/audio/filters/echo.c +++ /dev/null @@ -1,187 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * - * 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 . - * - */ - -#include -#include -#include -#include -#include -#include "rarch_dsp.h" - -// 4 source echo. - -#ifndef ALIGNED -#ifdef __GNUC__ -#define ALIGNED __attribute__((aligned(16))) -#else -#define ALIGNED -#endif -#endif - -#ifndef min -#define min(a, b) (((a) < (b)) ? (a) : (b)) -#endif - -struct echo_filter -{ - float *history; // history buffer - int pos; // current position in history buffer - int amp; // amplification of echoes (0-256) - int delay; // delay in number of samples - int ms; // delay in miliseconds - int rate; // sample rate - - float f_amp; // amplification (0-1) - float input_rate; -}; - -struct echo_filter_data -{ - struct echo_filter echo_l; - struct echo_filter echo_r; - float buf[4096]; -}; - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init echo_dsp_plugin_init -#endif - -static float echo_process(void *data, float in) -{ - struct echo_filter *echo = (struct echo_filter*)data; - - float smp = echo->history[echo->pos]; - smp *= echo->f_amp; - smp += in; - echo->history[echo->pos] = smp; - echo->pos = (echo->pos + 1) % echo->delay; - return smp; -} - -static void echo_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - int num_samples, i; - struct echo_filter_data *echo = (struct echo_filter_data*)data; - output->samples = echo->buf; - num_samples = input->frames * 2; - - for (i = 0; i < num_samples;) - { - echo->buf[i] = echo_process(&echo->echo_l, input->samples[i]); - i++; - echo->buf[i] = echo_process(&echo->echo_r, input->samples[i]); - i++; - } - output->frames = input->frames; -} - -static void echo_dsp_free(void *data) -{ - struct echo_filter_data *echo = (struct echo_filter_data*)data; - - if (echo) - free(echo); -} - -static void echo_set_delay(void *data, int ms) -{ - int new_delay, how_much, i; - float *new_history; - struct echo_filter *echo = (struct echo_filter*)data; - - new_delay = ms * echo->input_rate / 1000; - if (new_delay == 0) - new_delay = 1; - - new_history = (float*)malloc(new_delay * sizeof(float)); - memset(new_history, 0, new_delay * sizeof(float)); - - if (echo->history) - { - how_much = echo->delay - echo->pos; - how_much = min(how_much, new_delay); - memcpy(new_history, echo->history + echo->pos, how_much * sizeof(float)); - - if (how_much < new_delay) - { - i = how_much; - how_much = new_delay - how_much; - how_much = min(how_much, echo->delay); - how_much = min(how_much, echo->pos); - memcpy(new_history + i, echo->history, how_much * sizeof(float)); - } - - if (echo->history) - free(echo->history); - } - echo->history = new_history; - echo->pos = 0; - echo->delay = new_delay; - echo->ms = ms; -} - -static void *echo_dsp_init(const rarch_dsp_info_t *info) -{ - struct echo_filter_data *echo = (struct echo_filter_data*)calloc(1, sizeof(*echo));; - - if (!echo) - return NULL; - - echo->echo_l.history = NULL; - echo->echo_l.input_rate = info->input_rate; - echo_set_delay(&echo->echo_l, 200); - echo->echo_l.amp = 128; - echo->echo_l.f_amp = (float)echo->echo_l.amp / 256.0f; - echo->echo_l.pos = 0; - - echo->echo_r.history = NULL; - echo->echo_r.input_rate = info->input_rate; - echo_set_delay(&echo->echo_r, 200); - echo->echo_r.amp = 128; - echo->echo_r.f_amp = (float)echo->echo_r.amp / 256.0f; - echo->echo_r.pos = 0; - - fprintf(stderr, "[Echo] loaded!\n"); - - return echo; -} - -static void echo_dsp_config(void *data) -{ - (void)data; -} - -static const struct dspfilter_implementation generic_echo_dsp = { - echo_dsp_init, - echo_dsp_process, - echo_dsp_free, - RARCH_DSP_API_VERSION, - echo_dsp_config, - "Echo", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_echo_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/eq.c b/audio/filters/eq.c deleted file mode 100644 index c76160c73a..0000000000 --- a/audio/filters/eq.c +++ /dev/null @@ -1,422 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include //FIXME: This is a dependency missing pretty much everywhere except Linux -#include -#include -#include -#include "boolean.h" -#include - -#ifndef M_PI -#define M_PI 3.14159265 -#endif - -#ifndef EQ_COEFF_SIZE -#define EQ_COEFF_SIZE 256 -#endif - -#ifndef EQ_FILT_SIZE -#define EQ_FILT_SIZE (EQ_COEFF_SIZE * 2) -#endif - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init eq_dsp_plugin_init -#endif - -typedef struct dsp_eq_state dsp_eq_state_t; - -static complex float phase_lut[2 * EQ_FILT_SIZE + 1]; -static complex float * const phase_lut_ptr = phase_lut + EQ_FILT_SIZE; - -static void generate_phase_lut(void) -{ - int i; - for (i = -EQ_FILT_SIZE; i <= EQ_FILT_SIZE; i++) - { - float phase = (float)i / EQ_FILT_SIZE; - phase_lut_ptr[i] = cexpf(M_PI * I * phase); - } -} - -static inline unsigned bitrange(unsigned len) -{ - unsigned ret; - ret = 0; - while ((len >>= 1)) - ret++; - - return ret; -} - -static inline unsigned bitswap(unsigned i, unsigned range) -{ - unsigned ret, shifts; - ret = 0; - for (shifts = 0; shifts < range; shifts++) - ret |= i & (1 << (range - shifts - 1)) ? (1 << shifts) : 0; - - return ret; -} - -// When interleaving the butterfly buffer, addressing puts bits in reverse. -// [0, 1, 2, 3, 4, 5, 6, 7] => [0, 4, 2, 6, 1, 5, 3, 7] -static void interleave(complex float *butterfly_buf, size_t samples) -{ - unsigned range, i, target; - range = bitrange(samples); - for (i = 0; i < samples; i++) - { - target = bitswap(i, range); - if (target > i) - { - complex float tmp = butterfly_buf[target]; - butterfly_buf[target] = butterfly_buf[i]; - butterfly_buf[i] = tmp; - } - } -} - -static void butterfly(complex float *a, complex float *b, complex float mod) -{ - complex float a_, b_; - mod *= *b; - a_ = *a + mod; - b_ = *a - mod; - *a = a_; - *b = b_; -} - -static void butterflies(complex float *butterfly_buf, int phase_dir, size_t step_size, size_t samples) -{ - unsigned i, j; - int phase_step; - for (i = 0; i < samples; i += 2 * step_size) - { - phase_step = EQ_FILT_SIZE * phase_dir / (int)step_size; - for (j = i; j < i + step_size; j++) - butterfly(&butterfly_buf[j], &butterfly_buf[j + step_size], phase_lut_ptr[phase_step * (int)(j - i)]); - } -} - -static void calculate_fft_butterfly(complex float *butterfly_buf, size_t samples) -{ - unsigned step_size; - // Interleave buffer to work with FFT. - interleave(butterfly_buf, samples); - - // Fly, lovely butterflies! :D - for (step_size = 1; step_size < samples; step_size *= 2) - butterflies(butterfly_buf, -1, step_size, samples); -} - -static void calculate_fft(const float *data, complex float *butterfly_buf, size_t samples) -{ - unsigned i, step_size; - for (i = 0; i < samples; i++) - butterfly_buf[i] = data[i]; - - // Interleave buffer to work with FFT. - interleave(butterfly_buf, samples); - - // Fly, lovely butterflies! :D - for (step_size = 1; step_size < samples; step_size *= 2) - butterflies(butterfly_buf, -1, step_size, samples); -} - -static void calculate_ifft(complex float *butterfly_buf, size_t samples) -{ - unsigned step_size, i; - float factor; - - // Interleave buffer to work with FFT. - interleave(butterfly_buf, samples); - - // Fly, lovely butterflies! In opposite direction! :D - for (step_size = 1; step_size < samples; step_size *= 2) - butterflies(butterfly_buf, 1, step_size, samples); - - factor = 1.0 / samples; - for (i = 0; i < samples; i++) - butterfly_buf[i] *= factor; -} - -struct eq_band -{ - float gain; - unsigned min_bin; - unsigned max_bin; -}; - -struct dsp_eq_state -{ - struct eq_band *bands; - unsigned num_bands; - - complex float fft_coeffs[EQ_FILT_SIZE]; - float cosine_window[EQ_COEFF_SIZE]; - - float last_buf[EQ_COEFF_SIZE]; - float stage_buf[EQ_FILT_SIZE]; - unsigned stage_ptr; -}; - -static void calculate_band_range(struct eq_band *band, float norm_freq) -{ - unsigned max_bin = (unsigned)round(norm_freq * EQ_COEFF_SIZE); - - band->gain = 1.0; - band->max_bin = max_bin; -} - -static void recalculate_fft_filt(dsp_eq_state_t *eq) -{ - unsigned i, j, start, end; - complex float freq_response[EQ_FILT_SIZE] = {0.0f}; - - for (i = 0; i < eq->num_bands; i++) - { - for (j = eq->bands[i].min_bin; j <= eq->bands[i].max_bin; j++) - freq_response[j] = eq->bands[i].gain; - } - - memset(eq->fft_coeffs, 0, sizeof(eq->fft_coeffs)); - - for (start = 1, end = EQ_COEFF_SIZE - 1; start < EQ_COEFF_SIZE / 2; start++, end--) - freq_response[end] = freq_response[start]; - - calculate_ifft(freq_response, EQ_COEFF_SIZE); - - // ifftshift(). Needs to be done for some reason ... TODO: Figure out why :D - memcpy(eq->fft_coeffs + EQ_COEFF_SIZE / 2, freq_response + 0, EQ_COEFF_SIZE / 2 * sizeof(complex float)); - memcpy(eq->fft_coeffs + 0, freq_response + EQ_COEFF_SIZE / 2, EQ_COEFF_SIZE / 2 * sizeof(complex float)); - - for (i = 0; i < EQ_COEFF_SIZE; i++) - eq->fft_coeffs[i] *= eq->cosine_window[i]; - - calculate_fft_butterfly(eq->fft_coeffs, EQ_FILT_SIZE); -} - -static void dsp_eq_free(dsp_eq_state_t *eq) -{ - if (eq) - { - if (eq->bands) - free(eq->bands); - free(eq); - } -} - -static dsp_eq_state_t *dsp_eq_new(float input_rate, const float *bands, unsigned num_bands) -{ - unsigned i; - dsp_eq_state_t *state; - - for (i = 1; i < num_bands; i++) - { - if (bands[i] <= bands[i - 1]) - return NULL; - } - - if (num_bands < 2) - return NULL; - - state = (dsp_eq_state_t*)calloc(1, sizeof(*state)); - if (!state) - return NULL; - - state->num_bands = num_bands; - - state->bands = (struct eq_band*)calloc(num_bands, sizeof(struct eq_band)); - if (!state->bands) - goto error; - - calculate_band_range(&state->bands[0], ((bands[0] + bands[1]) / 2.0) / input_rate); - state->bands[0].min_bin = 0; - - for (i = 1; i < num_bands - 1; i++) - { - calculate_band_range(&state->bands[i], ((bands[i + 1] + bands[i + 0]) / 2.0) / input_rate); - state->bands[i].min_bin = state->bands[i - 1].max_bin + 1; - - if (state->bands[i].max_bin < state->bands[i].min_bin) - fprintf(stderr, "[Equalizer]: Band @ %.2f Hz does not have enough spectral resolution to fit.\n", bands[i]); - } - - state->bands[num_bands - 1].max_bin = EQ_COEFF_SIZE / 2; - state->bands[num_bands - 1].min_bin = state->bands[num_bands - 2].max_bin + 1; - state->bands[num_bands - 1].gain = 1.0f; - - for (i = 0; i < EQ_COEFF_SIZE; i++) - state->cosine_window[i] = cosf(M_PI * (i + 0.5 - EQ_COEFF_SIZE / 2) / EQ_COEFF_SIZE); - - generate_phase_lut(); - recalculate_fft_filt(state); - - return state; - -error: - dsp_eq_free(state); - return NULL; -} - -#if 0 -static void dsp_eq_set_gain(dsp_eq_state_t *eq, unsigned band, float gain) -{ - assert(band < eq->num_bands); - - eq->bands[band].gain = gain; - recalculate_fft_filt(eq); -} -#endif - -static size_t dsp_eq_process(dsp_eq_state_t *eq, float *output, size_t out_samples, - const float *input, size_t in_samples, unsigned stride) -{ - size_t written = 0; - while (in_samples) - { - unsigned i; - size_t to_read = EQ_COEFF_SIZE - eq->stage_ptr; - - if (to_read > in_samples) - to_read = in_samples; - - for (i = 0; i < to_read; i++, input += stride) - eq->stage_buf[eq->stage_ptr + i] = *input; - - in_samples -= to_read; - eq->stage_ptr += to_read; - - if (eq->stage_ptr >= EQ_COEFF_SIZE) - { - complex float butterfly_buf[EQ_FILT_SIZE]; - if (out_samples < EQ_COEFF_SIZE) - return written; - - calculate_fft(eq->stage_buf, butterfly_buf, EQ_FILT_SIZE); - for (i = 0; i < EQ_FILT_SIZE; i++) - butterfly_buf[i] *= eq->fft_coeffs[i]; - - calculate_ifft(butterfly_buf, EQ_FILT_SIZE); - - for (i = 0; i < EQ_COEFF_SIZE; i++, output += stride, out_samples--, written++) - *output = crealf(butterfly_buf[i]) + eq->last_buf[i]; - - for (i = 0; i < EQ_COEFF_SIZE; i++) - eq->last_buf[i] = crealf(butterfly_buf[i + EQ_COEFF_SIZE]); - - eq->stage_ptr = 0; - } - } - - return written; -} - - -#if 0 -static float db2gain(float val) -{ - return powf(10.0, val / 20.0); -} - -static float noise(void) -{ - return 2.0 * ((float)(rand()) / RAND_MAX - 0.5); -} -#endif - -struct equalizer_filter_data -{ - dsp_eq_state_t *eq_l; - dsp_eq_state_t *eq_r; - float out_buffer[8092]; -}; - -static size_t equalizer_process(void *data, const float *in, unsigned frames) -{ - struct equalizer_filter_data *eq = (struct equalizer_filter_data*)data; - - size_t written = dsp_eq_process(eq->eq_l, eq->out_buffer + 0, 4096, in + 0, frames, 2); - dsp_eq_process(eq->eq_r, eq->out_buffer + 1, 4096, in + 1, frames, 2); - - return written; -} - -static void * eq_dsp_init(const rarch_dsp_info_t *info) -{ - const float bands[] = { 30, 80, 150, 250, 500, 800, 1000, 2000, 3000, 5000, 8000, 10000, 12000, 15000 }; - struct equalizer_filter_data *eq = (struct equalizer_filter_data*)calloc(1, sizeof(*eq)); - - if (!eq) - return NULL; - - eq->eq_l = dsp_eq_new(info->input_rate, bands, sizeof(bands) / sizeof(bands[0])); - eq->eq_r = dsp_eq_new(info->input_rate, bands, sizeof(bands) / sizeof(bands[0])); - - return eq; -} - -static void eq_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - struct equalizer_filter_data *eq = (struct equalizer_filter_data*)data; - - output->samples = eq->out_buffer; - size_t out_frames = equalizer_process(eq, input->samples, input->frames); - output->frames = out_frames; -} - -static void eq_dsp_free(void *data) -{ - struct equalizer_filter_data *eq = (struct equalizer_filter_data*)data; - - if (eq) - { - dsp_eq_free(eq->eq_l); - dsp_eq_free(eq->eq_r); - free(eq); - } -} - -static void eq_dsp_config(void *data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_eq_dsp = { - eq_dsp_init, - eq_dsp_process, - eq_dsp_free, - RARCH_DSP_API_VERSION, - eq_dsp_config, - "Equalizer", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_eq_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/iir.c b/audio/filters/iir.c deleted file mode 100644 index 335589bd53..0000000000 --- a/audio/filters/iir.c +++ /dev/null @@ -1,421 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * Copyright (C) 2012-2014 - Brad Miller - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include -#include -#include - -#ifdef __SSE2__ -#include -#endif - -#ifndef M_PI -#define M_PI 3.1415926535897932384626433832795 -#endif - -#ifndef ALIGNED -#ifdef __GNUC__ -#define ALIGNED __attribute__((aligned(16))); -#else -#define ALIGNED -#endif -#endif - -#define sqr(a) ((a) * (a)) - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init iir_dsp_plugin_init -#endif - -struct iir_filter -{ -#ifdef __SSE2__ - __m128 fir_coeff[2]; - __m128 fir_buf[2]; - - __m128 iir_coeff; - __m128 iir_buf; -#endif - float pf_freq, pf_qfact, pf_gain; - int type, pf_q_is_bandwidth; - float xn1,xn2,yn1,yn2; - float omega, cs, a1pha, beta, b0, b1, b2, a0, a1,a2, A, sn; -}; - -struct iir_filter_data -{ - struct iir_filter iir_l ALIGNED; - struct iir_filter iir_r ALIGNED; - float buf[4096] ALIGNED; - int rate; - unsigned type; -}; - -/* filter types */ -enum -{ - LPF, /* low pass filter */ - HPF, /* High pass filter */ - BPCSGF,/* band pass filter 1 */ - BPZPGF,/* band pass filter 2 */ - APF, /* Allpass filter*/ - NOTCH, /* Notch Filter */ - RIAA_phono, /* RIAA record/tape deemphasis */ - PEQ, /* Peaking band EQ filter */ - BBOOST, /* Bassboost filter */ - LSH, /* Low shelf filter */ - HSH, /* High shelf filter */ - RIAA_CD /* CD de-emphasis */ -}; - -//lynched from SoX >w> -static void iir_make_poly_from_roots(double const * roots, size_t num_roots, float * poly) -{ - size_t i, j; - poly[0] = 1; - poly[1] = -roots[0]; - memset(poly + 2, 0, (num_roots + 1 - 2) * sizeof(*poly)); - for (i = 1; i < num_roots; ++i) - for (j = num_roots; j > 0; --j) - poly[j] -= poly[j - 1] * roots[i]; -} - -static void iir_init(void *data, int samplerate, int filter_type) -{ - struct iir_filter *iir = (struct iir_filter*)data; - - if (!iir) - return; - - iir->xn1=0; - iir->xn2=0; - iir->yn1=0; - iir->yn2=0; - iir->omega = 2 * M_PI * iir->pf_freq/samplerate; - iir->cs = cos(iir->omega); - iir->sn = sin(iir->omega); - iir->a1pha = iir->sn / (2.0 * iir->pf_qfact); - iir->A = exp(log(10.0) * iir->pf_gain / 40); - iir->beta = sqrt(iir->A + iir->A); - //Set up filter coefficients according to type - switch (filter_type) - { - case LPF: - iir->b0 = (1.0 - iir->cs) / 2.0 ; - iir->b1 = 1.0 - iir->cs ; - iir->b2 = (1.0 - iir->cs) / 2.0 ; - iir->a0 = 1.0 + iir->a1pha ; - iir->a1 = -2.0 * iir->cs ; - iir->a2 = 1.0 - iir->a1pha ; - break; - case HPF: - iir->b0 = (1.0 + iir->cs) / 2.0 ; - iir->b1 = -(1.0 + iir->cs) ; - iir->b2 = (1.0 + iir->cs) / 2.0 ; - iir->a0 = 1.0 + iir->a1pha ; - iir->a1 = -2.0 * iir->cs ; - iir->a2 = 1.0 - iir->a1pha ; - break; - case APF: - iir->b0 = 1.0 - iir->a1pha; - iir->b1 = -2.0 * iir->cs; - iir->b2 = 1.0 + iir->a1pha; - iir->a0 = 1.0 + iir->a1pha; - iir->a1 = -2.0 * iir->cs; - iir->a2 = 1.0 - iir->a1pha; - break; - case BPZPGF: - iir->b0 = iir->a1pha ; - iir->b1 = 0.0 ; - iir->b2 = -iir->a1pha ; - iir->a0 = 1.0 + iir->a1pha ; - iir->a1 = -2.0 * iir->cs ; - iir->a2 = 1.0 - iir->a1pha ; - break; - case BPCSGF: - iir->b0=iir->sn/2.0; - iir->b1=0.0; - iir->b2=-iir->sn/2; - iir->a0=1.0+iir->a1pha; - iir->a1=-2.0*iir->cs; - iir->a2=1.0-iir->a1pha; - break; - case NOTCH: - iir->b0 = 1; - iir->b1 = -2 * iir->cs; - iir->b2 = 1; - iir->a0 = 1 + iir->a1pha; - iir->a1 = -2 * iir->cs; - iir->a2 = 1 - iir->a1pha; - break; - case RIAA_phono: /* http://www.dsprelated.com/showmessage/73300/3.php */ - if (samplerate == 44100) { - static const double zeros[] = {-0.2014898, 0.9233820}; - static const double poles[] = {0.7083149, 0.9924091}; - iir_make_poly_from_roots(zeros, (size_t)2, &iir->b0); - iir_make_poly_from_roots(poles, (size_t)2, &iir->a0); - } - else if (samplerate == 48000) { - static const double zeros[] = {-0.1766069, 0.9321590}; - static const double poles[] = {0.7396325, 0.9931330}; - iir_make_poly_from_roots(zeros, (size_t)2, &iir->b0); - iir_make_poly_from_roots(poles, (size_t)2, &iir->a0); - } - else if (samplerate == 88200) { - static const double zeros[] = {-0.1168735, 0.9648312}; - static const double poles[] = {0.8590646, 0.9964002}; - iir_make_poly_from_roots(zeros, (size_t)2, &iir->b0); - iir_make_poly_from_roots(poles, (size_t)2, &iir->a0); - } - else if (samplerate == 96000) { - static const double zeros[] = {-0.1141486, 0.9676817}; - static const double poles[] = {0.8699137, 0.9966946}; - iir_make_poly_from_roots(zeros, (size_t)2, &iir->b0); - iir_make_poly_from_roots(poles, (size_t)2, &iir->a0); - } - { /* Normalise to 0dB at 1kHz (Thanks to Glenn Davis) */ - double y = 2 * M_PI * 1000 / samplerate ; - double b_re = iir->b0 + iir->b1 * cos(-y) + iir->b2 * cos(-2 * y); - double a_re = iir->a0 + iir->a1 * cos(-y) + iir->a2 * cos(-2 * y); - double b_im = iir->b1 * sin(-y) + iir->b2 * sin(-2 * y); - double a_im = iir->a1 * sin(-y) + iir->a2 * sin(-2 * y); - double g = 1 / sqrt((sqr(b_re) + sqr(b_im)) / (sqr(a_re) + sqr(a_im))); - iir->b0 *= g; - iir->b1 *= g; - iir->b2 *= g; - } - break; - case PEQ: - iir->b0 = 1 + iir->a1pha * iir->A ; - iir->b1 = -2 * iir->cs ; - iir->b2 = 1 - iir->a1pha * iir->A ; - iir->a0 = 1 + iir->a1pha / iir->A ; - iir->a1 = -2 * iir->cs ; - iir->a2 = 1 - iir->a1pha / iir->A ; - break; - case BBOOST: - iir->beta = sqrt((iir->A * iir->A + 1) / 1.0 - (pow((iir->A - 1), 2))); - iir->b0 = iir->A * ((iir->A + 1) - (iir->A - 1) * iir->cs + iir->beta * iir->sn); - iir->b1 = 2 * iir->A * ((iir->A - 1) - (iir->A + 1) * iir->cs); - iir->b2 = iir->A * ((iir->A + 1) - (iir->A - 1) * iir->cs - iir->beta * iir->sn); - iir->a0 = ((iir->A + 1) + (iir->A - 1) * iir->cs + iir->beta * iir->sn); - iir->a1 = -2 * ((iir->A - 1) + (iir->A + 1) * iir->cs); - iir->a2 = (iir->A + 1) + (iir->A - 1) * iir->cs - iir->beta * iir->sn; - break; - case LSH: - iir->b0 = iir->A * ((iir->A + 1) - (iir->A - 1) * iir->cs + iir->beta * iir->sn); - iir->b1 = 2 * iir->A * ((iir->A - 1) - (iir->A + 1) * iir->cs); - iir->b2 = iir->A * ((iir->A + 1) - (iir->A - 1) * iir->cs - iir->beta * iir->sn); - iir->a0 = (iir->A + 1) + (iir->A - 1) * iir->cs + iir->beta * iir->sn; - iir->a1 = -2 * ((iir->A - 1) + (iir->A + 1) * iir->cs); - iir->a2 = (iir->A + 1) + (iir->A - 1) * iir->cs - iir->beta * iir->sn; - break; - case RIAA_CD: - iir->omega = 2 * M_PI * 5283 / samplerate; - iir->cs = cos(iir->omega); - iir->sn = sin(iir->omega); - iir->a1pha = iir->sn / (2.0 * 0.4845); - iir->A = exp(log(10.0) * -9.477 / 40); - iir->beta = sqrt(iir->A + iir->A); - case HSH: - iir->b0 = iir->A * ((iir->A + 1) + (iir->A - 1) * iir->cs + iir->beta * iir->sn); - iir->b1 = -2 * iir->A * ((iir->A - 1) + (iir->A + 1) * iir->cs); - iir->b2 = iir->A * ((iir->A + 1) + (iir->A - 1) * iir->cs - iir->beta * iir->sn); - iir->a0 = (iir->A + 1) - (iir->A - 1) * iir->cs + iir->beta * iir->sn; - iir->a1 = 2 * ((iir->A - 1) - (iir->A + 1) * iir->cs); - iir->a2 = (iir->A + 1) - (iir->A - 1) * iir->cs - iir->beta * iir->sn; - break; - default: - break; - } - -#ifdef __SSE2__ - iir->fir_coeff[0] = _mm_set_ps(iir->b1 / iir->a0, iir->b1 / iir->a0, iir->b0 / iir->a0, iir->b0 / iir->a0); - iir->fir_coeff[1] = _mm_set_ps(0.0f, 0.0f, iir->b2 / iir->a0, iir->b2 / iir->a0); - iir->iir_coeff = _mm_set_ps(-iir->a2 / iir->a0, -iir->a2 / iir->a0, -iir->a1 / iir->a0, -iir->a1 / iir->a0); -#endif -} - -#ifdef __SSE2__ -static void iir_process_batch(void *data, float *out, const float *in, unsigned frames) -{ - unsigned i; - struct iir_filter *iir = (struct iir_filter*)data; - - __m128 fir_coeff[2] = { iir->fir_coeff[0], iir->fir_coeff[1] }; - __m128 iir_coeff = iir->iir_coeff; - __m128 fir_buf[2] = { iir->fir_buf[0], iir->fir_buf[1] }; - __m128 iir_buf = iir->iir_buf; - - for (i = 0; (i + 4) <= (2 * frames); in += 4, i += 4, out += 4) - { - __m128 input = _mm_loadu_ps(in); - - fir_buf[1] = _mm_shuffle_ps(fir_buf[0], fir_buf[1], _MM_SHUFFLE(1, 0, 3, 2)); - fir_buf[0] = _mm_shuffle_ps(input, fir_buf[0], _MM_SHUFFLE(1, 0, 1, 0)); - - __m128 res[3] = { - _mm_mul_ps(fir_buf[0], fir_coeff[0]), - _mm_mul_ps(fir_buf[1], fir_coeff[1]), - _mm_mul_ps(iir_buf, iir_coeff), - }; - - __m128 result = _mm_add_ps(_mm_add_ps(res[0], res[1]), res[2]); - result = _mm_add_ps(result, _mm_shuffle_ps(result, result, _MM_SHUFFLE(0, 0, 3, 2))); - - iir_buf = _mm_shuffle_ps(result, iir_buf, _MM_SHUFFLE(1, 0, 1, 0)); - - fir_buf[1] = _mm_shuffle_ps(fir_buf[0], fir_buf[1], _MM_SHUFFLE(1, 0, 3, 2)); - fir_buf[0] = _mm_shuffle_ps(input, fir_buf[0], _MM_SHUFFLE(1, 0, 3, 2)); - - res[0] = _mm_mul_ps(fir_buf[0], fir_coeff[0]); - res[1] = _mm_mul_ps(fir_buf[1], fir_coeff[1]); - res[2] = _mm_mul_ps(iir_buf, iir_coeff); - - __m128 result2 = _mm_add_ps(_mm_add_ps(res[0], res[1]), res[2]); - result2 = _mm_add_ps(result2, _mm_shuffle_ps(result2, result2, _MM_SHUFFLE(0, 0, 3, 2))); - - iir_buf = _mm_shuffle_ps(result2, iir_buf, _MM_SHUFFLE(1, 0, 1, 0)); - - _mm_store_ps(out, _mm_shuffle_ps(result, result2, _MM_SHUFFLE(1, 0, 1, 0))); - } - - iir->fir_buf[0] = fir_buf[0]; - iir->fir_buf[1] = fir_buf[1]; - iir->iir_buf = iir_buf; -} -#endif - -static float iir_process(void *data, float samp) -{ - struct iir_filter *iir = (struct iir_filter*)data; - - float out, in = 0; - in = samp; - out = (iir->b0 * in + iir->b1 * iir->xn1 + iir->b2 * iir->xn2 - iir->a1 * iir->yn1 - iir->a2 * iir->yn2) / iir->a0; - iir->xn2 = iir->xn1; - iir->xn1 = in; - iir->yn2 = iir->yn1; - iir->yn1 = out; - return out; -} - -static void * iir_dsp_init(const rarch_dsp_info_t *info) -{ - struct iir_filter_data *iir = (struct iir_filter_data*)calloc(1, sizeof(*iir)); - - if (!iir) - return NULL; - - iir->rate = info->input_rate; - iir->type = 0; - iir->iir_l.pf_freq = 1024; - iir->iir_l.pf_qfact = 0.707; - iir->iir_l.pf_gain = 0; - iir_init(&iir->iir_l, info->input_rate, 0); - iir->iir_r.pf_freq = 1024; - iir->iir_r.pf_qfact = 0.707; - iir->iir_r.pf_gain = 0; - iir_init(&iir->iir_r, info->input_rate, 0); - - return iir; -} - -static void iir_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - int i, num_samples; - struct iir_filter_data *iir = (struct iir_filter_data*)data; - - output->samples = iir->buf; - - num_samples = input->frames * 2; - for (i = 0; ibuf[i] = iir_process(&iir->iir_l, input->samples[i]); - i++; - iir->buf[i] = iir_process(&iir->iir_r, input->samples[i]); - i++; - } - - output->frames = input->frames; -} - -#ifdef __SSE2__ -static void iir_dsp_process_sse2(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - struct iir_filter_data *iir = (struct iir_filter_data*)data; - - output->samples = iir->buf; - iir_process_batch(&iir->iir_l, iir->buf, input->samples, input->frames); - output->frames = input->frames; -} -#endif - -static void iir_dsp_free(void *data) -{ - struct iir_filter_data *iir = (struct iir_filter_data*)data; - - if (iir) - free(iir); -} - -static void iir_dsp_config(void* data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_iir_dsp = { - iir_dsp_init, - iir_dsp_process, - iir_dsp_free, - RARCH_DSP_API_VERSION, - iir_dsp_config, - "IIR", - NULL -}; - -#ifdef __SSE2__ -const struct dspfilter_implementation sse2_iir_dsp = { - iir_dsp_init, - iir_dsp_process_sse2, - iir_dsp_free, - RARCH_DSP_API_VERSION, - iir_dsp_config, - "IIR (SSE2)", - NULL -}; -#endif - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; -#ifdef __SSE2__ - if (simd & DSPFILTER_SIMD_SSE2) - return &sse2_iir_dsp; -#endif - return &generic_iir_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/phaser.c b/audio/filters/phaser.c deleted file mode 100644 index e5a211761a..0000000000 --- a/audio/filters/phaser.c +++ /dev/null @@ -1,193 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * Copyright (C) 2012-2014 - Brad Miller - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include -#include - -#ifndef M_PI -#define M_PI 3.1415926535897932384626433832795 -#endif - -#define PHASERLFOSHAPE 4.0 -#define PHASER_LFOSKIPSAMPLES 20 - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init phaser_dsp_plugin_init -#endif - -struct phaser_filter -{ - float freq; - float startphase; - float fb; - int depth; - int stages; - int drywet; - unsigned long skipcount; - float old[24]; - float gain; - float fbout; - float lfoskip; - float phase; -}; - -struct phaser_filter_data -{ - struct phaser_filter phase_l; - struct phaser_filter phase_r; - float buf[4096]; -}; - -static void phaser_init(void *data, int samplerate) -{ - int j; - struct phaser_filter *phaser = (struct phaser_filter*)data; - - phaser->skipcount = 0; - phaser->gain = 0.0; - phaser->fbout = 0.0; - phaser->lfoskip = phaser->freq * 2 * M_PI / samplerate; - phaser->phase = phaser->startphase * M_PI / 180; - for (j = 0; j < phaser->stages; j++) - phaser->old[j] = 0; -} - -static float phaser_process(void *data, float in) -{ - float m, tmp, out; - int j; - struct phaser_filter *phaser = (struct phaser_filter*)data; - - m = in + phaser->fbout * phaser->fb / 100; - - if (((phaser->skipcount++) % PHASER_LFOSKIPSAMPLES) == 0) - { - phaser->gain = (1 + cos(phaser->skipcount * phaser->lfoskip + phaser->phase)) / 2; - phaser->gain =(exp(phaser->gain * PHASERLFOSHAPE) - 1) / (exp(PHASERLFOSHAPE)-1); - phaser->gain = 1 - phaser->gain / 255 * phaser->depth; - } - for (j = 0; j < phaser->stages; j++) - { - tmp = phaser->old[j]; - phaser->old[j] = phaser->gain * tmp + m; - m = tmp - phaser->gain * phaser->old[j]; - } - phaser->fbout = m; - out = (m * phaser->drywet + in * (255 - phaser->drywet)) / 255; - if (out < -1.0) out = -1.0; - if (out > 1.0) out = 1.0; - return out; -} - -static void * phaser_dsp_init(const rarch_dsp_info_t *info) -{ - float freq, startphase, fb; - int depth, stages, drywet; - struct phaser_filter_data *phaser; - - freq = 0.4; - startphase = 0; - fb = 0; - depth = 100; - stages = 2; - drywet = 128; - - phaser = (struct phaser_filter_data*)calloc(1, sizeof(*phaser)); - - if (!phaser) - return NULL; - - phaser->phase_l.freq = freq; - phaser->phase_l.startphase = startphase; - phaser->phase_l.fb = fb; - phaser->phase_l.depth = depth; - phaser->phase_l.stages = stages; - phaser->phase_l.drywet = drywet; - phaser_init(&phaser->phase_l, info->input_rate); - - phaser->phase_r.freq = freq; - phaser->phase_r.startphase = startphase; - phaser->phase_r.fb = fb; - phaser->phase_r.depth = depth; - phaser->phase_r.stages = stages; - phaser->phase_r.drywet = drywet; - phaser_init(&phaser->phase_r, info->input_rate); - - return phaser; -} - -static void phaser_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - int i, num_samples; - struct phaser_filter_data *phaser = (struct phaser_filter_data*)data; - - output->samples = phaser->buf; - num_samples = input->frames * 2; - for (i = 0; ibuf[i] = phaser_process(&phaser->phase_l, input->samples[i]); - i++; - phaser->buf[i] = phaser_process(&phaser->phase_r, input->samples[i]); - i++; - } - output->frames = input->frames; -} - -static void phaser_dsp_free(void *data) -{ - struct phaser_filter_data *phaser = (struct phaser_filter_data*)data; - - if (phaser) - { - int j; - for (j = 0; j < phaser->phase_l.stages; j++) - phaser->phase_l.old[j] = 0; - for (j = 0; j < phaser->phase_r.stages; j++) - phaser->phase_r.old[j] = 0; - free(phaser); - } -} - -static void phaser_dsp_config(void *data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_phaser_dsp = { - phaser_dsp_init, - phaser_dsp_process, - phaser_dsp_free, - RARCH_DSP_API_VERSION, - phaser_dsp_config, - "Phaser", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_phaser_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/reverb.c b/audio/filters/reverb.c deleted file mode 100644 index 4c7e1062b7..0000000000 --- a/audio/filters/reverb.c +++ /dev/null @@ -1,397 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * Copyright (C) 2012-2014 - Brad Miller - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include - -#define NUMCOMBS 8 -#define NUMALLPASSES 4 -#define MUTED 0 -#define FIXEDGAIN 0.015f -#define SCALEWET 3 -#define SCALEDRY 2 -#define SCALEDAMP 0.4f -#define SCALEROOM 0.28f -#define OFFSETROOM 0.7f -#define INITIALROOM 0.5f -#define INITIALDAMP 0.5f -#define INITIALWET (1 / SCALEWET) -#define INITIALDRY 0 -#define INITIALWIDTH 1 -#define INITIALMODE 0 -#define FREEZEMODE 0.5f - -#define COMBTUNINGL1 1116 -#define COMBTUNINGL2 1188 -#define COMBTUNINGL3 1277 -#define COMBTUNINGL4 1356 -#define COMBTUNINGL5 1422 -#define COMBTUNINGL6 1491 -#define COMBTUNINGL7 1557 -#define COMBTUNINGL8 1617 -#define ALLPASSTUNINGL1 556 -#define ALLPASSTUNINGL2 441 -#define ALLPASSTUNINGL3 341 -#define ALLPASSTUNINGL4 225 - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init reverb_dsp_plugin_init -#endif - -struct comb -{ - float feedback; - float filterstore; - float damp1; - float damp2; - float *buffer; - int bufsize; - int bufidx; -}; - -struct allpass -{ - float feedback; - float *buffer; - int bufsize; - int bufidx; -}; - -struct revmodel -{ - float gain; - float roomsize, roomsize1; - float damp, damp1; - float wet, wet1, wet2; - float dry; - float width; - float mode; - - struct comb combL[NUMCOMBS]; - - struct allpass allpassL[NUMALLPASSES]; - - float bufcombL1[COMBTUNINGL1]; - float bufcombL2[COMBTUNINGL2]; - float bufcombL3[COMBTUNINGL3]; - float bufcombL4[COMBTUNINGL4]; - float bufcombL5[COMBTUNINGL5]; - float bufcombL6[COMBTUNINGL6]; - float bufcombL7[COMBTUNINGL7]; - float bufcombL8[COMBTUNINGL8]; - - float bufallpassL1[ALLPASSTUNINGL1]; - float bufallpassL2[ALLPASSTUNINGL2]; - float bufallpassL3[ALLPASSTUNINGL3]; - float bufallpassL4[ALLPASSTUNINGL4]; -}; - -// FIXME: Fix this really ugly hack -static inline float undenormalise(void *sample) -{ - if (((*(unsigned int*)sample) & 0x7f800000) == 0) - return 0.0f; - return *(float*)sample; -} - -static inline float comb_process(void *data, float input) -{ - struct comb *comb = (struct comb*)data; - float output; - - output = comb->buffer[comb->bufidx]; - undenormalise(&output); - - comb->filterstore = (output * comb->damp2) + (comb->filterstore * comb->damp1); - undenormalise(&comb->filterstore); - - comb->buffer[comb->bufidx] = input + (comb->filterstore * comb->feedback); - - if (++comb->bufidx >= comb->bufsize) - comb->bufidx = 0; - - return output; -} - - -static inline float allpass_process(void *data, float input) -{ - struct allpass *allpass = (struct allpass*)data; - float output, bufout; - - bufout = allpass->buffer[allpass->bufidx]; - undenormalise(&bufout); - - output = -input + bufout; - allpass->buffer[allpass->bufidx] = input + (bufout * allpass->feedback); - - if (++allpass->bufidx >= allpass->bufsize) - allpass->bufidx = 0; - - return output; -} - -static float revmodel_getmode(float mode) -{ - if (mode >= FREEZEMODE) - return 1; - else - return 0; -} - -static void revmodel_update(void *data) -{ - int i; - struct revmodel *rev = (struct revmodel*)data; - - rev->wet1 = rev->wet * (rev->width / 2 + 0.5f); - - if (rev->mode >= FREEZEMODE) - { - rev->roomsize1 = 1; - rev->damp1 = 0; - rev->gain = MUTED; - } - else - { - rev->roomsize1 = rev->roomsize; - rev->damp1 = rev->damp; - rev->gain = FIXEDGAIN; - } - - for (i = 0; i < NUMCOMBS; i++) - rev->combL[i].feedback = rev->roomsize1; - - for (i = 0; i < NUMCOMBS; i++) - { - rev->combL[i].damp1 = rev->damp1; - rev->combL[i].damp2 = 1 - rev->damp1; - } -} - -static void revmodel_set(void *data, float drytime, - float wettime, float damping, float roomwidth, float roomsize) -{ - int i, j; - struct revmodel *rev = (struct revmodel*)data; - - rev->wet = wettime; - revmodel_update(rev); - - rev->roomsize = roomsize; - revmodel_update(rev); - - rev->dry = drytime; - - rev->damp = damping; - revmodel_update(rev); - - rev->width = roomwidth; - revmodel_update(rev); - - rev->mode = INITIALMODE; - revmodel_update(rev); - - if (revmodel_getmode(rev->mode) >= FREEZEMODE) - return; - - for (i = 0; i < NUMCOMBS; i++) - { - for (j = 0; j < rev->combL[i].bufsize; j++) - rev->combL[i].buffer[j] = 0; - } - - for (i = 0; i < NUMALLPASSES; i++) - { - for (j = 0; j < rev->allpassL[i].bufsize; j++) - rev->allpassL[i].buffer[j] = 0; - } -} - -static void revmodel_init(void *data) -{ - struct revmodel *rev = (struct revmodel*)data; - - rev->combL[0].filterstore = 0; - rev->combL[0].bufidx = 0; - rev->combL[0].buffer = (float*)rev->bufcombL1; - rev->combL[0].bufsize = COMBTUNINGL1; - rev->combL[1].filterstore = 0; - rev->combL[1].bufidx = 0; - rev->combL[1].buffer = (float*)rev->bufcombL2; - rev->combL[1].bufsize = COMBTUNINGL2; - rev->combL[2].filterstore = 0; - rev->combL[2].bufidx = 0; - rev->combL[2].buffer = (float*)rev->bufcombL3; - rev->combL[2].bufsize = COMBTUNINGL3; - rev->combL[3].filterstore = 0; - rev->combL[3].bufidx = 0; - rev->combL[3].buffer = (float*)rev->bufcombL4; - rev->combL[3].bufsize = COMBTUNINGL4; - rev->combL[4].filterstore = 0; - rev->combL[4].bufidx = 0; - rev->combL[4].buffer = (float*)rev->bufcombL5; - rev->combL[4].bufsize = COMBTUNINGL5; - rev->combL[5].filterstore = 0; - rev->combL[5].bufidx = 0; - rev->combL[5].buffer = (float*)rev->bufcombL6; - rev->combL[5].bufsize = COMBTUNINGL6; - rev->combL[6].filterstore = 0; - rev->combL[6].bufidx = 0; - rev->combL[6].buffer = (float*)rev->bufcombL7; - rev->combL[6].bufsize = COMBTUNINGL7; - rev->combL[7].filterstore = 0; - rev->combL[7].bufidx = 0; - rev->combL[7].buffer = (float*)rev->bufcombL8; - rev->combL[7].bufsize = COMBTUNINGL8; - - rev->allpassL[0].bufidx = 0; - rev->allpassL[0].buffer = (float*)rev->bufallpassL1; - rev->allpassL[0].bufsize = ALLPASSTUNINGL1; - rev->allpassL[0].feedback = 0.5f; - rev->allpassL[1].bufidx = 0; - rev->allpassL[1].buffer = (float*)rev->bufallpassL2; - rev->allpassL[1].bufsize = ALLPASSTUNINGL2; - rev->allpassL[1].feedback = 0.5f; - rev->allpassL[2].bufidx = 0; - rev->allpassL[2].buffer = (float*)rev->bufallpassL3; - rev->allpassL[2].bufsize = ALLPASSTUNINGL3; - rev->allpassL[2].feedback = 0.5f; - rev->allpassL[3].bufidx = 0; - rev->allpassL[3].buffer = (float*)rev->bufallpassL4; - rev->allpassL[3].bufsize = ALLPASSTUNINGL4; - rev->allpassL[3].feedback = 0.5f; - -} - -static float revmodel_process(void *data, float in) -{ - float samp, mono_out, mono_in, input; - int i; - struct revmodel *rev = (struct revmodel*)data; - - samp = in; - mono_out = 0.0f; - mono_in = samp; - input = (mono_in) * rev->gain; - - for(i=0; i < NUMCOMBS; i++) - mono_out += comb_process(&rev->combL[i], input); - for(i = 0; i < NUMALLPASSES; i++) - mono_out = allpass_process(&rev->allpassL[i], mono_out); - samp = mono_in * rev->dry + mono_out * rev->wet1; - return samp; -} - - -#define REVMODEL_GETWET(revmodel) (revmodel->wet / SCALEWET) -#define REVMODEL_GETROOMSIZE(revmodel) ((revmodel->roomsize - OFFSETROOM) / SCALEROOM) -#define REVMODEL_GETDRY(revmodel) (revmodel->dry / SCALEDRY) -#define REVMODEL_GETWIDTH(revmodel) (revmodel->width) - - - -struct reverb_filter_data -{ - struct revmodel rev_l; - struct revmodel rev_r; - float buf[4096]; -}; - - -static void * reverb_dsp_init(const rarch_dsp_info_t *info) -{ - float drytime, wettime, damping, roomwidth, roomsize; - (void)info; - - drytime = 0.43; - wettime = 0.57; - damping = 0.45; - roomwidth = 0.56; - roomsize = 0.56; - - struct reverb_filter_data *reverb = (struct reverb_filter_data*)calloc(1, sizeof(*reverb)); - - if (!reverb) - return NULL; - - revmodel_init(&reverb->rev_l); - revmodel_set(&reverb->rev_l, INITIALDRY, - INITIALWET * SCALEWET, INITIALDAMP * SCALEDAMP, INITIALWIDTH, (INITIALROOM * SCALEROOM) + OFFSETROOM); - revmodel_set(&reverb->rev_l, drytime, wettime, damping, roomwidth, roomsize); - - revmodel_init(&reverb->rev_r); - revmodel_set(&reverb->rev_r, INITIALDRY, - INITIALWET * SCALEWET, INITIALDAMP * SCALEDAMP, INITIALWIDTH, (INITIALROOM * SCALEROOM) + OFFSETROOM); - revmodel_set(&reverb->rev_r, drytime, wettime, damping, roomwidth, roomsize); - - return reverb; -} - -static void reverb_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - int i, num_samples; - struct reverb_filter_data *reverb = (struct reverb_filter_data*)data; - - output->samples = reverb->buf; - num_samples = input->frames * 2; - for (i = 0; i < num_samples;) - { - reverb->buf[i] = revmodel_process(&reverb->rev_l, input->samples[i]); - i++; - reverb->buf[i] = revmodel_process(&reverb->rev_r, input->samples[i]); - i++; - } - output->frames = input->frames; -} - -static void reverb_dsp_free(void *data) -{ - struct reverb_filter_data *rev = (struct reverb_filter_data*)data; - - if (rev) - free(rev); -} - -static void reverb_dsp_config(void *data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_reverb_dsp = { - reverb_dsp_init, - reverb_dsp_process, - reverb_dsp_free, - RARCH_DSP_API_VERSION, - reverb_dsp_config, - "Reverberatation", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_reverb_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/volume.c b/audio/filters/volume.c deleted file mode 100644 index d95119b673..0000000000 --- a/audio/filters/volume.c +++ /dev/null @@ -1,131 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init volume_dsp_plugin_init -#endif - -struct volume_filter_data -{ -#ifdef __GNUC__ - float buf[4096] __attribute__((aligned(16))); -#else - float buf[4096]; -#endif - float m_vol; - float m_pan_vol_l; - float m_pan_vol_r; -}; - -#if 0 -static void pan2gain(float &left, float &right, int val) -{ - left = (100 - val) / 100.0f; - right = (val + 100) / 100.0f; - if (left > 1.0) - left = 1.0; - if (right > 1.0) - right = 1.0; -} - -static float db2gain(float val) -{ - return powf(10.0, val / 20.0); -} -#endif - -static void volume_process(void *data, const float *in, unsigned frames) -{ - float vol_left, vol_right; - unsigned i; - struct volume_filter_data *vol = (struct volume_filter_data*)data; - - if (!vol) - return; - - vol_left = vol->m_vol * vol->m_pan_vol_l; - vol_right = vol->m_vol * vol->m_pan_vol_r; - - for (i = 0; i < frames; i++) - { - vol->buf[(i << 1) + 0] = in[(i << 1) + 0] * vol_left; - vol->buf[(i << 1) + 1] = in[(i << 1) + 1] * vol_right; - } -} - -static void *volume_dsp_init(const rarch_dsp_info_t *info) -{ - struct volume_filter_data *vol = (struct volume_filter_data*)calloc(1, sizeof(*vol)); - (void)info; - - if (!vol) - return NULL; - - vol->m_vol = 1.0; - vol->m_pan_vol_l = 1.0; - vol->m_pan_vol_r = 1.0; - - return vol; -} - -static void volume_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - struct volume_filter_data *vol = (struct volume_filter_data*)data; - - output->samples = vol->buf; - volume_process(vol, input->samples, input->frames); - output->frames = input->frames; -} - -static void volume_dsp_free(void *data) -{ - struct volume_filter_data *vol = (struct volume_filter_data*)data; - - if (vol) - free(vol); -} - -static void volume_dsp_config(void *data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_volume_dsp = { - volume_dsp_init, - volume_dsp_process, - volume_dsp_free, - RARCH_DSP_API_VERSION, - volume_dsp_config, - "Volume", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_volume_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/audio/filters/wah.c b/audio/filters/wah.c deleted file mode 100644 index 485822bbb7..0000000000 --- a/audio/filters/wah.c +++ /dev/null @@ -1,187 +0,0 @@ -/* RetroArch - A frontend for libretro. - * Copyright (C) 2010-2014 - Hans-Kristian Arntzen - * Copyright (C) 2011-2014 - Daniel De Matteis - * Copyright (C) 2012-2014 - Brad Miller - * - * 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 . - * - */ - -#include "rarch_dsp.h" -#include -#include -#include -#include - -#ifndef M_PI -#define M_PI 3.1415926535897932384626433832795 -#endif - -#ifndef LFOSKIPSAMPLES -#define LFOSKIPSAMPLES 30 -#endif - -#ifdef RARCH_INTERNAL -#define rarch_dsp_plugin_init wah_dsp_plugin_init -#endif - -struct wahwah_filter -{ - float phase; - float lfoskip; - unsigned long skipcount; - float xn1, xn2, yn1, yn2; - float b0, b1, b2, a0, a1, a2; - float freq, startphase; - float depth, freqofs, res; -}; - -struct wahwah_filter_data -{ - struct wahwah_filter wah_l; - struct wahwah_filter wah_r; - float buf[4096]; -}; - -static void wahwah_init(void *data, int samplerate) -{ - struct wahwah_filter *wah = (struct wahwah_filter*)data; - - wah->lfoskip = wah->freq * 2 * M_PI / samplerate; - wah->skipcount = 0; - wah->xn1 = 0; - wah->xn2 = 0; - wah->yn1 = 0; - wah->yn2 = 0; - wah->b0 = 0; - wah->b1 = 0; - wah->b2 = 0; - wah->a0 = 0; - wah->a1 = 0; - wah->a2 = 0; - wah->phase = wah->startphase * M_PI / 180; -} - -static float wahwah_process(void *data, float samp) -{ - float frequency, omega, sn, cs, alpha; - float in, out; - struct wahwah_filter *wah = (struct wahwah_filter*)data; - - in = samp; - if ((wah->skipcount++) % LFOSKIPSAMPLES == 0) - { - frequency = (1 + cos(wah->skipcount * wah->lfoskip + wah->phase)) / 2; - frequency = frequency * wah->depth * (1 - wah->freqofs) + wah->freqofs; - frequency = exp((frequency - 1) * 6); - omega = M_PI * frequency; - sn = sin(omega); - cs = cos(omega); - alpha = sn / (2 * wah->res); - wah->b0 = (1 - cs) / 2; - wah->b1 = 1 - cs; - wah->b2 = (1 - cs) / 2; - wah->a0 = 1 + alpha; - wah->a1 = -2 * cs; - wah->a2 = 1 - alpha; - } - - out = (wah->b0 * in + wah->b1 * wah->xn1 + wah->b2 * wah->xn2 - wah->a1 * wah->yn1 - wah->a2 * wah->yn2) / wah->a0; - wah->xn2 = wah->xn1; - wah->xn1 = in; - wah->yn2 = wah->yn1; - wah->yn1 = out; - samp = out; - return samp; -} - -static void * wah_dsp_init(const rarch_dsp_info_t *info) -{ - float freq = 1.5; - float startphase = 0.0; - float res = 2.5; - float depth = 0.70; - float freqofs = 0.30; - - struct wahwah_filter_data *wah = (struct wahwah_filter_data*)calloc(1, sizeof(*wah)); - - if (!wah) - return NULL; - - wah->wah_l.depth = depth; - wah->wah_l.freqofs = freqofs; - wah->wah_l.freq = freq; - wah->wah_l.startphase = startphase; - wah->wah_l.res = res; - wahwah_init(&wah->wah_l, info->input_rate); - - wah->wah_r.depth = depth; - wah->wah_r.freqofs = freqofs; - wah->wah_r.freq = freq; - wah->wah_r.startphase = startphase; - wah->wah_r.res = res; - wahwah_init(&wah->wah_r, info->input_rate); - - return wah; -} - -static void wah_dsp_process(void *data, rarch_dsp_output_t *output, - const rarch_dsp_input_t *input) -{ - int num_samples, i; - struct wahwah_filter_data *wah = (struct wahwah_filter_data*)data; - - output->samples = wah->buf; - num_samples = input->frames * 2; - - for (i = 0; i < num_samples;) - { - wah->buf[i] = wahwah_process(&wah->wah_l, input->samples[i]); - i++; - wah->buf[i] = wahwah_process(&wah->wah_r, input->samples[i]); - i++; - } - output->frames = input->frames; -} - -static void wah_dsp_free(void *data) -{ - struct wahwah_filter_data *wah = (struct wahwah_filter_data*)data; - - if (wah) - free(wah); -} - -static void wah_dsp_config(void *data) -{ - (void)data; -} - -const struct dspfilter_implementation generic_wah_dsp = { - wah_dsp_init, - wah_dsp_process, - wah_dsp_free, - RARCH_DSP_API_VERSION, - wah_dsp_config, - "Wah", - NULL -}; - -const struct dspfilter_implementation *rarch_dsp_plugin_init(dspfilter_simd_mask_t simd) -{ - (void)simd; - return &generic_wah_dsp; -} - -#ifdef RARCH_INTERNAL -#undef rarch_dsp_plugin_init -#endif diff --git a/driver.h b/driver.h index ac0083e77b..a82e301c67 100644 --- a/driver.h +++ b/driver.h @@ -710,15 +710,7 @@ extern const struct softfilter_implementation *twoxbr_get_implementation(softfil extern const struct softfilter_implementation *darken_get_implementation(softfilter_simd_mask_t simd); extern const struct softfilter_implementation *scale2x_get_implementation(softfilter_simd_mask_t simd); -extern const struct dspfilter_implementation *echo_dsp_plugin_init(dspfilter_simd_mask_t simd); -#ifndef _WIN32 -extern const struct dspfilter_implementation *eq_dsp_plugin_init(dspfilter_simd_mask_t simd); -#endif -extern const struct dspfilter_implementation *iir_dsp_plugin_init(dspfilter_simd_mask_t simd); -extern const struct dspfilter_implementation *phaser_dsp_plugin_init(dspfilter_simd_mask_t simd); -extern const struct dspfilter_implementation *reverb_dsp_plugin_init(dspfilter_simd_mask_t simd); -extern const struct dspfilter_implementation *volume_dsp_plugin_init(dspfilter_simd_mask_t simd); -extern const struct dspfilter_implementation *wah_dsp_plugin_init(dspfilter_simd_mask_t simd); +// TODO: DSP plugs. #endif #include "driver_funcs.h" diff --git a/griffin/griffin.c b/griffin/griffin.c index e094822902..419323ad2a 100644 --- a/griffin/griffin.c +++ b/griffin/griffin.c @@ -485,17 +485,7 @@ FILTERS #include "../gfx/filters/lq2x.c" #include "../gfx/filters/phosphor2x.c" -#include "../audio/filters/echo.c" -#ifndef ANDROID -#ifndef _WIN32 -#include "../audio/filters/eq.c" -#endif -#endif -#include "../audio/filters/iir.c" -#include "../audio/filters/phaser.c" -#include "../audio/filters/reverb.c" -#include "../audio/filters/volume.c" -#include "../audio/filters/wah.c" +// TODO: Audio plugs. #endif /*============================================================ DYNAMIC @@ -503,6 +493,7 @@ DYNAMIC #include "../dynamic.c" #include "../dynamic_dummy.c" #include "../gfx/filter.c" +#include "../audio/dsp_filter.c" /*============================================================