avfilter: add audio frequency and phase shift filters
This commit is contained in:
Paul B Mahol
parent
d60927b676
commit
637c154a50
@ -36,6 +36,7 @@ version <next>:
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- AVS3 video decoder via libuavs3d
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- Cintel RAW decoder
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- VDPAU accelerated VP9 10/12bit decoding
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- afreqshift and aphaseshift filters
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version 4.3:
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@ -1314,6 +1314,21 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
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aformat=sample_fmts=u8|s16:channel_layouts=stereo
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@end example
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@section afreqshift
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Apply frequency shift to input audio samples.
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The filter accepts the following options:
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@table @option
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@item shift
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Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
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Default value is 0.0.
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@end table
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@subsection Commands
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This filter supports the above option as @ref{commands}.
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@section agate
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A gate is mainly used to reduce lower parts of a signal. This kind of signal
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@ -2064,6 +2079,21 @@ It accepts the following values:
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@end table
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@end table
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@section aphaseshift
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Apply phase shift to input audio samples.
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The filter accepts the following options:
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@table @option
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@item shift
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Specify phase shift. Allowed range is from -1.0 to 1.0.
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Default value is 0.0.
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@end table
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@subsection Commands
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This filter supports the above option as @ref{commands}.
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@section apulsator
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Audio pulsator is something between an autopanner and a tremolo.
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@ -50,6 +50,7 @@ OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
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OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
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OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_AFREQSHIFT_FILTER) += af_afreqshift.o
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OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
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OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
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OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
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@ -68,6 +69,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
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OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
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OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
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OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
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OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
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OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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379
libavfilter/af_afreqshift.c
Normal file
379
libavfilter/af_afreqshift.c
Normal file
@ -0,0 +1,379 @@
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/*
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* Copyright (c) Paul B Mahol
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* Copyright (c) Laurent de Soras, 2005
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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/channel_layout.h"
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#include "libavutil/ffmath.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 NB_COEFS 16
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typedef struct AFreqShift {
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const AVClass *class;
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double shift;
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double c[NB_COEFS];
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int64_t in_samples;
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AVFrame *i1, *o1;
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AVFrame *i2, *o2;
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void (*filter_channel)(AVFilterContext *ctx,
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int nb_samples,
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int sample_rate,
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const double *src, double *dst,
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double *i1, double *o1,
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double *i2, double *o2);
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} AFreqShift;
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts = NULL;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static void pfilter_channel(AVFilterContext *ctx,
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int nb_samples,
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int sample_rate,
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const double *src, double *dst,
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double *i1, double *o1,
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double *i2, double *o2)
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{
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AFreqShift *s = ctx->priv;
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double *c = s->c;
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double shift = s->shift * M_PI;
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double cos_theta = cos(shift);
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double sin_theta = sin(shift);
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for (int n = 0; n < nb_samples; n++) {
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double xn1 = src[n], xn2 = src[n];
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double I, Q;
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for (int j = 0; j < NB_COEFS / 2; j++) {
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I = c[j] * (xn1 + o2[j]) - i2[j];
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i2[j] = i1[j];
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i1[j] = xn1;
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o2[j] = o1[j];
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o1[j] = I;
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xn1 = I;
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}
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for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
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Q = c[j] * (xn2 + o2[j]) - i2[j];
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i2[j] = i1[j];
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i1[j] = xn2;
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o2[j] = o1[j];
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o1[j] = Q;
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xn2 = Q;
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}
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Q = o2[NB_COEFS - 1];
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dst[n] = I * cos_theta - Q * sin_theta;
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}
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}
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static void ffilter_channel(AVFilterContext *ctx,
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int nb_samples,
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int sample_rate,
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const double *src, double *dst,
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double *i1, double *o1,
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double *i2, double *o2)
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{
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AFreqShift *s = ctx->priv;
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double *c = s->c;
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double ts = 1. / sample_rate;
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double shift = s->shift;
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int64_t N = s->in_samples;
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for (int n = 0; n < nb_samples; n++) {
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double xn1 = src[n], xn2 = src[n];
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double I, Q, theta;
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for (int j = 0; j < NB_COEFS / 2; j++) {
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I = c[j] * (xn1 + o2[j]) - i2[j];
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i2[j] = i1[j];
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i1[j] = xn1;
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o2[j] = o1[j];
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o1[j] = I;
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xn1 = I;
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}
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for (int j = NB_COEFS / 2; j < NB_COEFS; j++) {
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Q = c[j] * (xn2 + o2[j]) - i2[j];
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i2[j] = i1[j];
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i1[j] = xn2;
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o2[j] = o1[j];
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o1[j] = Q;
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xn2 = Q;
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}
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Q = o2[NB_COEFS - 1];
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theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.);
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dst[n] = I * cos(theta) - Q * sin(theta);
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}
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}
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static void compute_transition_param(double *K, double *Q, double transition)
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{
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double kksqrt, e, e2, e4, k, q;
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k = tan((1. - transition * 2.) * M_PI / 4.);
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k *= k;
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kksqrt = pow(1 - k * k, 0.25);
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e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
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e2 = e * e;
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e4 = e2 * e2;
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q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));
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*Q = q;
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*K = k;
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}
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static double ipowp(double x, int64_t n)
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{
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double z = 1.;
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while (n != 0) {
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if (n & 1)
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z *= x;
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n >>= 1;
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x *= x;
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}
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return z;
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}
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static double compute_acc_num(double q, int order, int c)
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{
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int64_t i = 0;
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int j = 1;
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double acc = 0.;
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double q_ii1;
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do {
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q_ii1 = ipowp(q, i * (i + 1));
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q_ii1 *= sin((i * 2 + 1) * c * M_PI / order) * j;
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acc += q_ii1;
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j = -j;
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i++;
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} while (fabs(q_ii1) > 1e-100);
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return acc;
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}
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static double compute_acc_den(double q, int order, int c)
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{
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int64_t i = 1;
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int j = -1;
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double acc = 0.;
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double q_i2;
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do {
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q_i2 = ipowp(q, i * i);
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q_i2 *= cos(i * 2 * c * M_PI / order) * j;
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acc += q_i2;
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j = -j;
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i++;
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} while (fabs(q_i2) > 1e-100);
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return acc;
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}
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static double compute_coef(int index, double k, double q, int order)
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{
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const int c = index + 1;
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const double num = compute_acc_num(q, order, c) * pow(q, 0.25);
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const double den = compute_acc_den(q, order, c) + 0.5;
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const double ww = num / den;
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const double wwsq = ww * ww;
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const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
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const double coef = (1 - x) / (1 + x);
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return coef;
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}
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static void compute_coefs(double *coef_arr, int nbr_coefs, double transition)
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{
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const int order = nbr_coefs * 2 + 1;
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double k, q;
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compute_transition_param(&k, &q, transition);
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for (int n = 0; n < nbr_coefs; n++)
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coef_arr[(n / 2) + (n & 1) * nbr_coefs / 2] = compute_coef(n, k, q, order);
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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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AFreqShift *s = ctx->priv;
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compute_coefs(s->c, NB_COEFS, 2. * 20. / inlink->sample_rate);
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s->i1 = ff_get_audio_buffer(inlink, NB_COEFS);
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s->o1 = ff_get_audio_buffer(inlink, NB_COEFS);
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s->i2 = ff_get_audio_buffer(inlink, NB_COEFS);
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s->o2 = ff_get_audio_buffer(inlink, NB_COEFS);
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if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
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return AVERROR(ENOMEM);
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if (!strcmp(ctx->filter->name, "afreqshift"))
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s->filter_channel = ffilter_channel;
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else
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s->filter_channel = pfilter_channel;
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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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AVFilterLink *outlink = ctx->outputs[0];
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AFreqShift *s = ctx->priv;
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AVFrame *out;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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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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for (int ch = 0; ch < in->channels; ch++) {
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s->filter_channel(ctx, in->nb_samples,
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in->sample_rate,
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(const double *)in->extended_data[ch],
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(double *)out->extended_data[ch],
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(double *)s->i1->extended_data[ch],
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(double *)s->o1->extended_data[ch],
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(double *)s->i2->extended_data[ch],
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(double *)s->o2->extended_data[ch]);
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}
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s->in_samples += in->nb_samples;
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if (out != in)
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av_frame_free(&in);
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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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AFreqShift *s = ctx->priv;
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av_frame_free(&s->i1);
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av_frame_free(&s->o1);
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av_frame_free(&s->i2);
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av_frame_free(&s->o2);
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}
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#define OFFSET(x) offsetof(AFreqShift, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption afreqshift_options[] = {
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{ "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(afreqshift);
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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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{ NULL }
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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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{ NULL }
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};
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AVFilter ff_af_afreqshift = {
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.name = "afreqshift",
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.description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(AFreqShift),
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.priv_class = &afreqshift_class,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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.process_command = ff_filter_process_command,
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};
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static const AVOption aphaseshift_options[] = {
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{ "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aphaseshift);
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AVFilter ff_af_aphaseshift = {
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.name = "aphaseshift",
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.description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(AFreqShift),
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.priv_class = &aphaseshift_class,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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.process_command = ff_filter_process_command,
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};
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@ -43,6 +43,7 @@ extern AVFilter ff_af_afftdn;
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extern AVFilter ff_af_afftfilt;
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extern AVFilter ff_af_afir;
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extern AVFilter ff_af_aformat;
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extern AVFilter ff_af_afreqshift;
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extern AVFilter ff_af_agate;
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extern AVFilter ff_af_aiir;
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extern AVFilter ff_af_aintegral;
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@ -61,6 +62,7 @@ extern AVFilter ff_af_anull;
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extern AVFilter ff_af_apad;
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extern AVFilter ff_af_aperms;
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extern AVFilter ff_af_aphaser;
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extern AVFilter ff_af_aphaseshift;
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extern AVFilter ff_af_apulsator;
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extern AVFilter ff_af_arealtime;
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extern AVFilter ff_af_aresample;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 87
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#define LIBAVFILTER_VERSION_MINOR 88
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#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
|
||||
|
||||
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Reference in New Issue
Block a user