496 lines
15 KiB
C
496 lines
15 KiB
C
/*
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* Copyright (c) 2019 The FFmpeg Project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#define MAX_OVERSAMPLE 64
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enum ASoftClipTypes {
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ASC_HARD = -1,
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ASC_TANH,
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ASC_ATAN,
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ASC_CUBIC,
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ASC_EXP,
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ASC_ALG,
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ASC_QUINTIC,
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ASC_SIN,
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ASC_ERF,
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NB_TYPES,
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};
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typedef struct Lowpass {
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float fb0, fb1, fb2;
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float fa0, fa1, fa2;
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double db0, db1, db2;
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double da0, da1, da2;
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} Lowpass;
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typedef struct ASoftClipContext {
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const AVClass *class;
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int type;
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int oversample;
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int64_t delay;
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double threshold;
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double output;
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double param;
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Lowpass lowpass[MAX_OVERSAMPLE];
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AVFrame *frame[2];
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void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
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int nb_samples, int channels, int start, int end);
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} ASoftClipContext;
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#define OFFSET(x) offsetof(ASoftClipContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption asoftclip_options[] = {
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{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
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{ "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, "types" },
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{ "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
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{ "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
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{ "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
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{ "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" },
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{ "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
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{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
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{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
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{ "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
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{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
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{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
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{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
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{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(asoftclip);
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static void get_lowpass(Lowpass *s,
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double frequency,
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double sample_rate)
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{
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double w0 = 2 * M_PI * frequency / sample_rate;
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double alpha = sin(w0) / (2 * 0.8);
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double factor;
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s->da0 = 1 + alpha;
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s->da1 = -2 * cos(w0);
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s->da2 = 1 - alpha;
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s->db0 = (1 - cos(w0)) / 2;
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s->db1 = 1 - cos(w0);
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s->db2 = (1 - cos(w0)) / 2;
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s->da1 /= s->da0;
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s->da2 /= s->da0;
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s->db0 /= s->da0;
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s->db1 /= s->da0;
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s->db2 /= s->da0;
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s->da0 /= s->da0;
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factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
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s->db0 *= factor;
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s->db1 *= factor;
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s->db2 *= factor;
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s->fa0 = s->da0;
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s->fa1 = s->da1;
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s->fa2 = s->da2;
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s->fb0 = s->db0;
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s->fb1 = s->db1;
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s->fb2 = s->db2;
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}
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static inline float run_lowpassf(const Lowpass *const s,
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float src, float *w)
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{
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float dst;
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dst = src * s->fb0 + w[0];
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w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
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w[1] = s->fb2 * src - s->fa2 * dst;
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return dst;
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}
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static void filter_flt(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels,
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int start, int end)
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{
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const int oversample = s->oversample;
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const int nb_osamples = nb_samples * oversample;
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const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
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float threshold = s->threshold;
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float gain = s->output * threshold;
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float factor = 1.f / threshold;
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float param = s->param;
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for (int c = start; c < end; c++) {
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float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
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const float *src = sptr[c];
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float *dst = dptr[c];
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for (int n = 0; n < nb_samples; n++) {
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dst[oversample * n] = src[n];
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for (int m = 1; m < oversample; m++)
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dst[oversample * n + m] = 0.f;
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}
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = tanhf(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= 1.5f)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sample - 0.1481f * powf(sample, 3.f);
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dst[n] *= gain;
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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dst[n] = sample / (sqrtf(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sample - 0.08192f * powf(sample, 5.f);
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dst[n] *= gain;
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_osamples; n++) {
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float sample = dst[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sinf(sample);
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dst[n] *= gain;
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = erff(dst[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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av_assert0(0);
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}
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w = (float *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
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for (int n = 0; n < nb_samples; n++)
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dst[n] = dst[n * oversample] * scale;
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}
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}
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static inline double run_lowpassd(const Lowpass *const s,
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double src, double *w)
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{
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double dst;
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dst = src * s->db0 + w[0];
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w[0] = s->db1 * src + w[1] - s->da1 * dst;
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w[1] = s->db2 * src - s->da2 * dst;
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return dst;
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}
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static void filter_dbl(ASoftClipContext *s,
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void **dptr, const void **sptr,
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int nb_samples, int channels,
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int start, int end)
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{
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const int oversample = s->oversample;
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const int nb_osamples = nb_samples * oversample;
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const double scale = oversample > 1 ? oversample * 0.5 : 1.;
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double threshold = s->threshold;
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double gain = s->output * threshold;
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double factor = 1. / threshold;
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double param = s->param;
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for (int c = start; c < end; c++) {
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double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
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const double *src = sptr[c];
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double *dst = dptr[c];
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for (int n = 0; n < nb_samples; n++) {
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dst[oversample * n] = src[n];
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for (int m = 1; m < oversample; m++)
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dst[oversample * n + m] = 0.f;
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}
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
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switch (s->type) {
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case ASC_HARD:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = av_clipd(dst[n] * factor, -1., 1.);
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dst[n] *= gain;
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}
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break;
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case ASC_TANH:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = tanh(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_ATAN:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
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dst[n] *= gain;
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}
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break;
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case ASC_CUBIC:
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= 1.5)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sample - 0.1481 * pow(sample, 3.);
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dst[n] *= gain;
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}
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break;
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case ASC_EXP:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
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dst[n] *= gain;
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}
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break;
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case ASC_ALG:
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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dst[n] = sample / (sqrt(param + sample * sample));
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dst[n] *= gain;
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}
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break;
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case ASC_QUINTIC:
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= 1.25)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sample - 0.08192 * pow(sample, 5.);
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dst[n] *= gain;
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}
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break;
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case ASC_SIN:
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for (int n = 0; n < nb_osamples; n++) {
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double sample = dst[n] * factor;
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if (FFABS(sample) >= M_PI_2)
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dst[n] = FFSIGN(sample);
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else
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dst[n] = sin(sample);
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dst[n] *= gain;
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}
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break;
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case ASC_ERF:
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for (int n = 0; n < nb_osamples; n++) {
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dst[n] = erf(dst[n] * factor);
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dst[n] *= gain;
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}
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break;
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default:
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av_assert0(0);
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}
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w = (double *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
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for (int n = 0; n < nb_osamples && oversample > 1; n++)
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dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
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for (int n = 0; n < nb_samples; n++)
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dst[n] = dst[n * oversample] * scale;
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}
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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ASoftClipContext *s = ctx->priv;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
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case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
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default: av_assert0(0);
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}
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s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
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s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
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if (!s->frame[0] || !s->frame[1])
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return AVERROR(ENOMEM);
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for (int i = 0; i < MAX_OVERSAMPLE; i++) {
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get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
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}
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return 0;
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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int nb_samples;
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int channels;
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} ThreadData;
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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ASoftClipContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *out = td->out;
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AVFrame *in = td->in;
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const int channels = td->channels;
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const int nb_samples = td->nb_samples;
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const int start = (channels * jobnr) / nb_jobs;
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const int end = (channels * (jobnr+1)) / nb_jobs;
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s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
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nb_samples, channels, start, end);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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ASoftClipContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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int nb_samples, channels;
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ThreadData td;
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AVFrame *out;
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if (av_frame_is_writable(in) && s->oversample == 1) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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nb_samples = in->nb_samples;
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channels = in->channels;
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td.in = in;
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td.out = out;
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td.nb_samples = nb_samples;
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td.channels = channels;
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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FFMIN(channels, ff_filter_get_nb_threads(ctx)));
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if (out != in)
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av_frame_free(&in);
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out->nb_samples /= s->oversample;
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ASoftClipContext *s = ctx->priv;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_asoftclip = {
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.name = "asoftclip",
|
|
.description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
|
|
.priv_size = sizeof(ASoftClipContext),
|
|
.priv_class = &asoftclip_class,
|
|
FILTER_INPUTS(inputs),
|
|
FILTER_OUTPUTS(outputs),
|
|
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
|
|
.uninit = uninit,
|
|
.process_command = ff_filter_process_command,
|
|
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|