ref: f8667d2d023388c96d0f2d4ac93ef8d15dca1d17
dir: /src/reverb.c/
/* libSoX effect: stereo reverberation * Copyright (c) 2007 robs@users.sourceforge.net * Filter design based on freeverb by Jezar at Dreampoint. * * This library is free software; you can redistribute it and/or modify it * under the terms of the GNU Lesser General Public License as published by * the Free Software Foundation; either version 2.1 of the License, or (at * your option) any later version. * * This library 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 Lesser * General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with this library; if not, write to the Free Software Foundation, * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "sox_i.h" #include "fifo.h" #define lsx_zalloc(var, n) var = lsx_calloc(n, sizeof(*var)) #define filter_create(p, n) (p)->ptr=lsx_zalloc((p)->buffer, (p)->size=(size_t)(n)) #define filter_advance(p) if (--(p)->ptr < (p)->buffer) (p)->ptr += (p)->size #define filter_delete(p) free((p)->buffer) typedef struct { size_t size; float * buffer, * ptr; float store; } filter_t; static float comb_process(filter_t * p, /* gcc -O2 will inline this */ float const * input, float const * feedback, float const * hf_damping) { float output = *p->ptr; p->store = output + (p->store - output) * *hf_damping; *p->ptr = *input + p->store * *feedback; filter_advance(p); return output; } static float allpass_process(filter_t * p, /* gcc -O2 will inline this */ float const * input) { float output = *p->ptr; *p->ptr = *input + output * .5; filter_advance(p); return output - *input; } static const size_t /* Filter delay lengths in samples (44100Hz sample-rate) */ comb_lengths[] = {1116, 1188, 1277, 1356, 1422, 1491, 1557, 1617}, allpass_lengths[] = {225, 341, 441, 556}; #define stereo_adjust 12 typedef struct { filter_t comb [array_length(comb_lengths)]; filter_t allpass[array_length(allpass_lengths)]; } filter_array_t; static void filter_array_create(filter_array_t * p, double rate, double scale, double offset) { size_t i; double r = rate * (1 / 44100.); /* Compensate for actual sample-rate */ for (i = 0; i < array_length(comb_lengths); ++i, offset = -offset) filter_create(&p->comb[i], scale * r * (comb_lengths[i] + stereo_adjust * offset) + .5); for (i = 0; i < array_length(allpass_lengths); ++i, offset = -offset) filter_create(&p->allpass[i], r * (allpass_lengths[i] + stereo_adjust * offset) + .5); } static void filter_array_process(filter_array_t * p, size_t length, float const * input, float * output, float const * feedback, float const * hf_damping, float const * gain) { while (length--) { float out = 0, in = *input++; size_t i = array_length(comb_lengths) - 1; do out += comb_process(p->comb + i, &in, feedback, hf_damping); while (i--); i = array_length(allpass_lengths) - 1; do out = allpass_process(p->allpass + i, &out); while (i--); *output++ = out * *gain; } } static void filter_array_delete(filter_array_t * p) { size_t i; for (i = 0; i < array_length(allpass_lengths); ++i) filter_delete(&p->allpass[i]); for (i = 0; i < array_length(comb_lengths); ++i) filter_delete(&p->comb[i]); } typedef struct { float feedback; float hf_damping; float gain; fifo_t input_fifo; filter_array_t chan[2]; float * out[2]; } reverb_t; static void reverb_create(reverb_t * p, double sample_rate_Hz, double wet_gain_dB, double room_scale, /* % */ double reverberance, /* % */ double hf_damping, /* % */ double pre_delay_ms, double stereo_depth, size_t buffer_size, float * * out) { size_t i, delay = pre_delay_ms / 1000 * sample_rate_Hz + .5; double scale = room_scale / 100 * .9 + .1; double depth = stereo_depth / 100; double a = -1 / log(1 - /**/.3 /**/); /* Set minimum feedback */ double b = 100 / (log(1 - /**/.98/**/) * a + 1); /* Set maximum feedback */ memset(p, 0, sizeof(*p)); p->feedback = 1 - exp((reverberance - b) / (a * b)); p->hf_damping = hf_damping / 100 * .3 + .2; p->gain = dB_to_linear(wet_gain_dB) * .015; fifo_create(&p->input_fifo, sizeof(float)); memset(fifo_write(&p->input_fifo, delay, 0), 0, delay * sizeof(float)); for (i = 0; i <= ceil(depth); ++i) { filter_array_create(p->chan + i, sample_rate_Hz, scale, i * depth); out[i] = lsx_zalloc(p->out[i], buffer_size); } } static void reverb_process(reverb_t * p, size_t length) { size_t i; for (i = 0; i < 2 && p->out[i]; ++i) filter_array_process(p->chan + i, length, (float *) fifo_read_ptr(&p->input_fifo), p->out[i], &p->feedback, &p->hf_damping, &p->gain); fifo_read(&p->input_fifo, length, NULL); } static void reverb_delete(reverb_t * p) { size_t i; for (i = 0; i < 2 && p->out[i]; ++i) { free(p->out[i]); filter_array_delete(p->chan + i); } fifo_delete(&p->input_fifo); } /*------------------------------- SoX Wrapper --------------------------------*/ typedef struct { double reverberance, hf_damping, pre_delay_ms; double stereo_depth, wet_gain_dB, room_scale; sox_bool wet_only; size_t ichannels, ochannels; struct { reverb_t reverb; float * dry, * wet[2]; } chan[2]; } priv_t; static int getopts(sox_effect_t * effp, int argc, char **argv) { priv_t * p = (priv_t *)effp->priv; p->reverberance = p->hf_damping = 50; /* Set non-zero defaults */ p->stereo_depth = p->room_scale = 100; --argc, ++argv; p->wet_only = argc && (!strcmp(*argv, "-w") || !strcmp(*argv, "--wet-only")) && (--argc, ++argv, sox_true); do { /* break-able block */ NUMERIC_PARAMETER(reverberance, 0, 100) NUMERIC_PARAMETER(hf_damping, 0, 100) NUMERIC_PARAMETER(room_scale, 0, 100) NUMERIC_PARAMETER(stereo_depth, 0, 100) NUMERIC_PARAMETER(pre_delay_ms, 0, 500) NUMERIC_PARAMETER(wet_gain_dB, -10, 10) } while (0); return argc ? lsx_usage(effp) : SOX_SUCCESS; } static int start(sox_effect_t * effp) { priv_t * p = (priv_t *)effp->priv; size_t i; p->ichannels = p->ochannels = 1; effp->out_signal.rate = effp->in_signal.rate; if (effp->in_signal.channels > 2 && p->stereo_depth) { lsx_warn("stereo-depth not applicable with >2 channels"); p->stereo_depth = 0; } if (effp->in_signal.channels == 1 && p->stereo_depth) effp->out_signal.channels = p->ochannels = 2; else effp->out_signal.channels = effp->in_signal.channels; if (effp->in_signal.channels == 2 && p->stereo_depth) p->ichannels = p->ochannels = 2; else effp->flows = effp->in_signal.channels; for (i = 0; i < p->ichannels; ++i) reverb_create( &p->chan[i].reverb, effp->in_signal.rate, p->wet_gain_dB, p->room_scale, p->reverberance, p->hf_damping, p->pre_delay_ms, p->stereo_depth, effp->global_info->global_info->bufsiz / p->ochannels, p->chan[i].wet); if (effp->in_signal.mult) *effp->in_signal.mult /= !p->wet_only + 2 * dB_to_linear(max(0,p->wet_gain_dB)); return SOX_SUCCESS; } static int flow(sox_effect_t * effp, const sox_sample_t * ibuf, sox_sample_t * obuf, size_t * isamp, size_t * osamp) { priv_t * p = (priv_t *)effp->priv; size_t c, i, w, len = min(*isamp / p->ichannels, *osamp / p->ochannels); SOX_SAMPLE_LOCALS; *isamp = len * p->ichannels, *osamp = len * p->ochannels; for (c = 0; c < p->ichannels; ++c) p->chan[c].dry = fifo_write(&p->chan[c].reverb.input_fifo, len, 0); for (i = 0; i < len; ++i) for (c = 0; c < p->ichannels; ++c) p->chan[c].dry[i] = SOX_SAMPLE_TO_FLOAT_32BIT(*ibuf++, effp->clips); for (c = 0; c < p->ichannels; ++c) reverb_process(&p->chan[c].reverb, len); if (p->ichannels == 2) for (i = 0; i < len; ++i) for (w = 0; w < 2; ++w) { float out = (1 - p->wet_only) * p->chan[w].dry[i] + .5 * (p->chan[0].wet[w][i] + p->chan[1].wet[w][i]); *obuf++ = SOX_FLOAT_32BIT_TO_SAMPLE(out, effp->clips); } else for (i = 0; i < len; ++i) for (w = 0; w < p->ochannels; ++w) { float out = (1 - p->wet_only) * p->chan[0].dry[i] + p->chan[0].wet[w][i]; *obuf++ = SOX_FLOAT_32BIT_TO_SAMPLE(out, effp->clips); } return SOX_SUCCESS; } static int stop(sox_effect_t * effp) { priv_t * p = (priv_t *)effp->priv; size_t i; for (i = 0; i < p->ichannels; ++i) reverb_delete(&p->chan[i].reverb); return SOX_SUCCESS; } sox_effect_handler_t const *lsx_reverb_effect_fn(void) { static sox_effect_handler_t handler = {"reverb", "[-w|--wet-only]" " [reverberance (50%)" " [HF-damping (50%)" " [room-scale (100%)" " [stereo-depth (100%)" " [pre-delay (0ms)" " [wet-gain (0dB)" "]]]]]]", SOX_EFF_MCHAN, getopts, start, flow, NULL, stop, NULL, sizeof(priv_t) }; return &handler; }