/* * Copyright (c) Paul B Mahol * Copyright (c) Laurent de Soras, 2005 * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "libavutil/channel_layout.h" #include "libavutil/ffmath.h" #include "libavutil/opt.h" #include "avfilter.h" #include "audio.h" #include "formats.h" #define MAX_NB_COEFFS 16 typedef struct AFreqShift { const AVClass *class; double shift; double level; int nb_coeffs; int old_nb_coeffs; double cd[MAX_NB_COEFFS * 2]; float cf[MAX_NB_COEFFS * 2]; int64_t in_samples; AVFrame *i1, *o1; AVFrame *i2, *o2; void (*filter_channel)(AVFilterContext *ctx, int channel, AVFrame *in, AVFrame *out); } AFreqShift; static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }; #define PFILTER(name, type, sin, cos, cc) \ static void pfilter_channel_## name(AVFilterContext *ctx, \ int ch, \ AVFrame *in, AVFrame *out) \ { \ AFreqShift *s = ctx->priv; \ const int nb_samples = in->nb_samples; \ const type *src = (const type *)in->extended_data[ch]; \ type *dst = (type *)out->extended_data[ch]; \ type *i1 = (type *)s->i1->extended_data[ch]; \ type *o1 = (type *)s->o1->extended_data[ch]; \ type *i2 = (type *)s->i2->extended_data[ch]; \ type *o2 = (type *)s->o2->extended_data[ch]; \ const int nb_coeffs = s->nb_coeffs; \ const type *c = s->cc; \ const type level = s->level; \ type shift = s->shift * M_PI; \ type cos_theta = cos(shift); \ type sin_theta = sin(shift); \ \ for (int n = 0; n < nb_samples; n++) { \ type xn1 = src[n], xn2 = src[n]; \ type I, Q; \ \ for (int j = 0; j < nb_coeffs; j++) { \ I = c[j] * (xn1 + o2[j]) - i2[j]; \ i2[j] = i1[j]; \ i1[j] = xn1; \ o2[j] = o1[j]; \ o1[j] = I; \ xn1 = I; \ } \ \ for (int j = nb_coeffs; j < nb_coeffs*2; j++) { \ Q = c[j] * (xn2 + o2[j]) - i2[j]; \ i2[j] = i1[j]; \ i1[j] = xn2; \ o2[j] = o1[j]; \ o1[j] = Q; \ xn2 = Q; \ } \ Q = o2[nb_coeffs * 2 - 1]; \ \ dst[n] = (I * cos_theta - Q * sin_theta) * level; \ } \ } PFILTER(flt, float, sin, cos, cf) PFILTER(dbl, double, sin, cos, cd) #define FFILTER(name, type, sin, cos, fmod, cc) \ static void ffilter_channel_## name(AVFilterContext *ctx, \ int ch, \ AVFrame *in, AVFrame *out) \ { \ AFreqShift *s = ctx->priv; \ const int nb_samples = in->nb_samples; \ const type *src = (const type *)in->extended_data[ch]; \ type *dst = (type *)out->extended_data[ch]; \ type *i1 = (type *)s->i1->extended_data[ch]; \ type *o1 = (type *)s->o1->extended_data[ch]; \ type *i2 = (type *)s->i2->extended_data[ch]; \ type *o2 = (type *)s->o2->extended_data[ch]; \ const int nb_coeffs = s->nb_coeffs; \ const type *c = s->cc; \ const type level = s->level; \ type ts = 1. / in->sample_rate; \ type shift = s->shift; \ int64_t N = s->in_samples; \ \ for (int n = 0; n < nb_samples; n++) { \ type xn1 = src[n], xn2 = src[n]; \ type I, Q, theta; \ \ for (int j = 0; j < nb_coeffs; j++) { \ I = c[j] * (xn1 + o2[j]) - i2[j]; \ i2[j] = i1[j]; \ i1[j] = xn1; \ o2[j] = o1[j]; \ o1[j] = I; \ xn1 = I; \ } \ \ for (int j = nb_coeffs; j < nb_coeffs*2; j++) { \ Q = c[j] * (xn2 + o2[j]) - i2[j]; \ i2[j] = i1[j]; \ i1[j] = xn2; \ o2[j] = o1[j]; \ o1[j] = Q; \ xn2 = Q; \ } \ Q = o2[nb_coeffs * 2 - 1]; \ \ theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.); \ dst[n] = (I * cos(theta) - Q * sin(theta)) * level; \ } \ } FFILTER(flt, float, sinf, cosf, fmodf, cf) FFILTER(dbl, double, sin, cos, fmod, cd) static void compute_transition_param(double *K, double *Q, double transition) { double kksqrt, e, e2, e4, k, q; k = tan((1. - transition * 2.) * M_PI / 4.); k *= k; kksqrt = pow(1 - k * k, 0.25); e = 0.5 * (1. - kksqrt) / (1. + kksqrt); e2 = e * e; e4 = e2 * e2; q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4))); *Q = q; *K = k; } static double ipowp(double x, int64_t n) { double z = 1.; while (n != 0) { if (n & 1) z *= x; n >>= 1; x *= x; } return z; } static double compute_acc_num(double q, int order, int c) { int64_t i = 0; int j = 1; double acc = 0.; double q_ii1; do { q_ii1 = ipowp(q, i * (i + 1)); q_ii1 *= sin((i * 2 + 1) * c * M_PI / order) * j; acc += q_ii1; j = -j; i++; } while (fabs(q_ii1) > 1e-100); return acc; } static double compute_acc_den(double q, int order, int c) { int64_t i = 1; int j = -1; double acc = 0.; double q_i2; do { q_i2 = ipowp(q, i * i); q_i2 *= cos(i * 2 * c * M_PI / order) * j; acc += q_i2; j = -j; i++; } while (fabs(q_i2) > 1e-100); return acc; } static double compute_coef(int index, double k, double q, int order) { const int c = index + 1; const double num = compute_acc_num(q, order, c) * pow(q, 0.25); const double den = compute_acc_den(q, order, c) + 0.5; const double ww = num / den; const double wwsq = ww * ww; const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq); const double coef = (1 - x) / (1 + x); return coef; } static void compute_coefs(double *coef_arrd, float *coef_arrf, int nbr_coefs, double transition) { const int order = nbr_coefs * 2 + 1; double k, q; compute_transition_param(&k, &q, transition); for (int n = 0; n < nbr_coefs; n++) { const int idx = (n / 2) + (n & 1) * nbr_coefs / 2; coef_arrd[idx] = compute_coef(n, k, q, order); coef_arrf[idx] = coef_arrd[idx]; } } static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AFreqShift *s = ctx->priv; if (s->old_nb_coeffs != s->nb_coeffs) compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate); s->old_nb_coeffs = s->nb_coeffs; s->i1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2); s->o1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2); s->i2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2); s->o2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2); if (!s->i1 || !s->o1 || !s->i2 || !s->o2) return AVERROR(ENOMEM); if (inlink->format == AV_SAMPLE_FMT_DBLP) { if (!strcmp(ctx->filter->name, "afreqshift")) s->filter_channel = ffilter_channel_dbl; else s->filter_channel = pfilter_channel_dbl; } else { if (!strcmp(ctx->filter->name, "afreqshift")) s->filter_channel = ffilter_channel_flt; else s->filter_channel = pfilter_channel_flt; } return 0; } typedef struct ThreadData { AVFrame *in, *out; } ThreadData; static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { AFreqShift *s = ctx->priv; ThreadData *td = arg; AVFrame *out = td->out; AVFrame *in = td->in; const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs; const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; for (int ch = start; ch < end; ch++) s->filter_channel(ctx, ch, in, out); return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; AVFilterLink *outlink = ctx->outputs[0]; AFreqShift *s = ctx->priv; AVFrame *out; ThreadData td; if (s->old_nb_coeffs != s->nb_coeffs) compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate); s->old_nb_coeffs = s->nb_coeffs; if (av_frame_is_writable(in)) { out = in; } else { out = ff_get_audio_buffer(outlink, in->nb_samples); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); } td.in = in; td.out = out; ff_filter_execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); s->in_samples += in->nb_samples; if (out != in) av_frame_free(&in); return ff_filter_frame(outlink, out); } static av_cold void uninit(AVFilterContext *ctx) { AFreqShift *s = ctx->priv; av_frame_free(&s->i1); av_frame_free(&s->o1); av_frame_free(&s->i2); av_frame_free(&s->o2); } #define OFFSET(x) offsetof(AFreqShift, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption afreqshift_options[] = { { "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS }, { "level", "set output level", OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS }, { "order", "set filter order", OFFSET(nb_coeffs),AV_OPT_TYPE_INT, {.i64=8}, 1, MAX_NB_COEFFS, FLAGS }, { NULL } }; AVFILTER_DEFINE_CLASS(afreqshift); static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .filter_frame = filter_frame, .config_props = config_input, }, }; static const AVFilterPad outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, }, }; const AVFilter ff_af_afreqshift = { .name = "afreqshift", .description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."), .priv_size = sizeof(AFreqShift), .priv_class = &afreqshift_class, .uninit = uninit, FILTER_INPUTS(inputs), FILTER_OUTPUTS(outputs), FILTER_SAMPLEFMTS_ARRAY(sample_fmts), .process_command = ff_filter_process_command, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC | AVFILTER_FLAG_SLICE_THREADS, }; static const AVOption aphaseshift_options[] = { { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS }, { "level", "set output level",OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS }, { "order", "set filter order",OFFSET(nb_coeffs), AV_OPT_TYPE_INT,{.i64=8}, 1, MAX_NB_COEFFS, FLAGS }, { NULL } }; AVFILTER_DEFINE_CLASS(aphaseshift); const AVFilter ff_af_aphaseshift = { .name = "aphaseshift", .description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."), .priv_size = sizeof(AFreqShift), .priv_class = &aphaseshift_class, .uninit = uninit, FILTER_INPUTS(inputs), FILTER_OUTPUTS(outputs), FILTER_SAMPLEFMTS_ARRAY(sample_fmts), .process_command = ff_filter_process_command, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC | AVFILTER_FLAG_SLICE_THREADS, };