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Commit fa04ec72 authored by Paul B Mahol's avatar Paul B Mahol
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avfilter: add afftfilter 

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent 56c182c4
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Changelog
View file @ fa04ec72
......@@ -56,6 +56,7 @@ version <next>:
- ahistogram filter
- only seek with the right mouse button in ffplay
- toggle full screen when double-clicking with the left mouse button in ffplay
- afftfilt filter
version 2.8:
......
configure
View file @ fa04ec72
......@@ -2841,6 +2841,8 @@ unix_protocol_deps="sys_un_h"
unix_protocol_select="network"
# filters
afftfilt_filter_deps="avcodec"
afftfilt_filter_select="fft"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
......@@ -6065,6 +6067,7 @@ done
enabled zlib && add_cppflags -DZLIB_CONST
# conditional library dependencies, in linking order
enabled afftfilt_filter && prepend avfilter_deps "avcodec"
enabled amovie_filter && prepend avfilter_deps "avformat avcodec"
enabled aresample_filter && prepend avfilter_deps "swresample"
enabled asyncts_filter && prepend avfilter_deps "avresample"
......
doc/filters.texi
View file @ fa04ec72
......@@ -733,6 +733,83 @@ afade=t=out:st=875:d=25
@end example
@end itemize
@section afftfilt
Apply arbitrary expressions to samples in frequency domain.
@table @option
@item real
Set frequency domain real expression for each separate channel separated
by '|'. Default is "1".
If the number of input channels is greater than the number of
expressions, the last specified expression is used for the remaining
output channels.
@item imag
Set frequency domain imaginary expression for each separate channel
separated by '|'. If not set, @var{real} option is used.
Each expression in @var{real} and @var{imag} can contain the following
constants:
@table @option
@item sr
sample rate
@item b
current frequency bin number
@item nb
number of available bins
@item ch
channel number of the current expression
@item chs
number of channels
@item pts
current frame pts
@end table
@item win_size
Set window size.
It accepts the following values:
@table @samp
@item w16
@item w32
@item w64
@item w128
@item w256
@item w512
@item w1024
@item w2048
@item w4096
@item w8192
@item w16384
@item w32768
@item w65536
@end table
Default is @code{w4096}
@item win_func
Set window function. Default is @code{hann}.
@item overlap
Set window overlap. If set to 1, the recommended overlap for selected
window function will be picked. Default is @code{0.75}.
@end table
@subsection Examples
@itemize
@item
Increase first 50 bins by 0.1 and lower all other frequencies by factor of 10:
@example
afftfilt="1.1*between(b\,0\,49)+0.1*between(b\,50\,f)"
@end example
@end itemize
@anchor{aformat}
@section aformat
......
libavfilter/Makefile
View file @ fa04ec72
......@@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o
OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
......
libavfilter/af_afftfilt.c 0 → 100644
View file @ fa04ec72
/*
* Copyright (c) 2016 Paul B Mahol
*
* 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/audio_fifo.h"
#include "libavutil/avstring.h"
#include "libavfilter/internal.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "libavutil/eval.h"
#include "audio.h"
#include "window_func.h"
typedef struct AFFTFiltContext {
const AVClass *class;
char *real_str;
char *img_str;
int fft_bits;
FFTContext *fft, *ifft;
FFTComplex **fft_data;
int nb_exprs;
int window_size;
AVExpr **real;
AVExpr **imag;
AVAudioFifo *fifo;
int64_t pts;
int hop_size;
float overlap;
AVFrame *buffer;
int start, end;
int win_func;
float win_scale;
float *window_func_lut;
} AFFTFiltContext;
static const char *const var_names[] = { "sr", "b", "nb", "ch", "chs", "pts", NULL };
enum { VAR_SAMPLE_RATE, VAR_BIN, VAR_NBBINS, VAR_CHANNEL, VAR_CHANNELS, VAR_PTS, VAR_VARS_NB };
#define OFFSET(x) offsetof(AFFTFiltContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption afftfilt_options[] = {
{ "real", "set channels real expressions", OFFSET(real_str), AV_OPT_TYPE_STRING, {.str = "1" }, 0, 0, A },
{ "imag", "set channels imaginary expressions", OFFSET(img_str), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, A },
{ "win_size", "set window size", OFFSET(fft_bits), AV_OPT_TYPE_INT, {.i64=12}, 4, 16, A, "fft" },
{ "w16", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "fft" },
{ "w32", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "fft" },
{ "w64", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "fft" },
{ "w128", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "fft" },
{ "w256", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, A, "fft" },
{ "w512", 0, 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, A, "fft" },
{ "w1024", 0, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, A, "fft" },
{ "w2048", 0, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, A, "fft" },
{ "w4096", 0, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, A, "fft" },
{ "w8192", 0, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, A, "fft" },
{ "w16384", 0, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, A, "fft" },
{ "w32768", 0, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, A, "fft" },
{ "w65536", 0, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, A, "fft" },
{ "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64 = WFUNC_HANNING}, 0, NB_WFUNC-1, A, "win_func" },
{ "rect", "Rectangular", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT}, 0, 0, A, "win_func" },
{ "bartlett", "Bartlett", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, A, "win_func" },
{ "hann", "Hann", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, A, "win_func" },
{ "hanning", "Hanning", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, A, "win_func" },
{ "hamming", "Hamming", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING}, 0, 0, A, "win_func" },
{ "sine", "Sine", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE}, 0, 0, A, "win_func" },
{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 1, A },
{ NULL },
};
AVFILTER_DEFINE_CLASS(afftfilt);
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AFFTFiltContext *s = ctx->priv;
char *saveptr = NULL;
int ret = 0, ch, i;
float overlap;
char *args, *last_expr = NULL;
s->fft = av_fft_init(s->fft_bits, 0);
s->ifft = av_fft_init(s->fft_bits, 1);
if (!s->fft || !s->ifft)
return AVERROR(ENOMEM);
s->window_size = 1 << s->fft_bits;
s->fft_data = av_calloc(inlink->channels, sizeof(*s->fft_data));
if (!s->fft_data)
return AVERROR(ENOMEM);
for (ch = 0; ch < inlink->channels; ch++) {
s->fft_data[ch] = av_calloc(s->window_size, sizeof(**s->fft_data));
if (!s->fft_data[ch])
return AVERROR(ENOMEM);
}
s->real = av_calloc(inlink->channels, sizeof(*s->real));
if (!s->real)
return AVERROR(ENOMEM);
s->imag = av_calloc(inlink->channels, sizeof(*s->imag));
if (!s->imag)
return AVERROR(ENOMEM);
args = av_strdup(s->real_str);
if (!args)
return AVERROR(ENOMEM);
for (ch = 0; ch < inlink->channels; ch++) {
char *arg = av_strtok(ch == 0 ? args : NULL, "|", &saveptr);
ret = av_expr_parse(&s->real[ch], arg ? arg : last_expr, var_names,
NULL, NULL, NULL, NULL, 0, ctx);
if (ret < 0)
break;
if (arg)
last_expr = arg;
s->nb_exprs++;
}
av_free(args);
args = av_strdup(s->img_str ? s->img_str : s->real_str);
if (!args)
return AVERROR(ENOMEM);
for (ch = 0; ch < inlink->channels; ch++) {
char *arg = av_strtok(ch == 0 ? args : NULL, "|", &saveptr);
ret = av_expr_parse(&s->imag[ch], arg ? arg : last_expr, var_names,
NULL, NULL, NULL, NULL, 0, ctx);
if (ret < 0)
break;
if (arg)
last_expr = arg;
}
av_free(args);
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
if (!s->fifo)
return AVERROR(ENOMEM);
s->window_func_lut = av_realloc_f(s->window_func_lut, s->window_size,
sizeof(*s->window_func_lut));
if (!s->window_func_lut)
return AVERROR(ENOMEM);
ff_generate_window_func(s->window_func_lut, s->window_size, s->win_func, &overlap);
if (s->overlap == 1)
s->overlap = overlap;
for (s->win_scale = 0, i = 0; i < s->window_size; i++) {
s->win_scale += s->window_func_lut[i] * s->window_func_lut[i];
}
s->hop_size = s->window_size * (1 - s->overlap);
if (s->hop_size <= 0)
return AVERROR(EINVAL);
s->buffer = ff_get_audio_buffer(inlink, s->window_size * 2);
if (!s->buffer)
return AVERROR(ENOMEM);
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AFFTFiltContext *s = ctx->priv;
const int window_size = s->window_size;
const float f = 1. / s->win_scale;
double values[VAR_VARS_NB];
AVFrame *out, *in = NULL;
int ch, n, ret, i, j, k;
int start = s->start, end = s->end;
av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
av_frame_free(&frame);
while (av_audio_fifo_size(s->fifo) >= window_size) {
if (!in) {
in = ff_get_audio_buffer(outlink, window_size);
if (!in)
return AVERROR(ENOMEM);
}
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
if (ret < 0)
break;
for (ch = 0; ch < inlink->channels; ch++) {
const float *src = (float *)in->extended_data[ch];
FFTComplex *fft_data = s->fft_data[ch];
for (n = 0; n < in->nb_samples; n++) {
fft_data[n].re = src[n] * s->window_func_lut[n];
fft_data[n].im = 0;
}
for (; n < window_size; n++) {
fft_data[n].re = 0;
fft_data[n].im = 0;
}
}
values[VAR_PTS] = s->pts;
values[VAR_SAMPLE_RATE] = inlink->sample_rate;
values[VAR_NBBINS] = window_size / 2;
values[VAR_CHANNELS] = inlink->channels;
for (ch = 0; ch < inlink->channels; ch++) {
FFTComplex *fft_data = s->fft_data[ch];
float *buf = (float *)s->buffer->extended_data[ch];
int x;
values[VAR_CHANNEL] = ch;
av_fft_permute(s->fft, fft_data);
av_fft_calc(s->fft, fft_data);
for (n = 0; n < window_size / 2; n++) {
float fr, fi;
values[VAR_BIN] = n;
fr = av_expr_eval(s->real[ch], values, s);
fi = av_expr_eval(s->imag[ch], values, s);
fft_data[n].re *= fr;
fft_data[n].im *= fi;
}
for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
fft_data[n].re = fft_data[x].re;
fft_data[n].im = -fft_data[x].im;
}
av_fft_permute(s->ifft, fft_data);
av_fft_calc(s->ifft, fft_data);
start = s->start;
end = s->end;
k = end;
for (i = 0, j = start; j < k && i < window_size; i++, j++) {
buf[j] += s->fft_data[ch][i].re * f;
}
for (; i < window_size; i++, j++) {
buf[j] = s->fft_data[ch][i].re * f;
}
start += s->hop_size;
end = j;
}
s->start = start;
s->end = end;
if (start >= window_size) {
float *dst, *buf;
start -= window_size;
end -= window_size;
s->start = start;
s->end = end;
out = ff_get_audio_buffer(outlink, window_size);
if (!out) {
ret = AVERROR(ENOMEM);
break;
}
out->pts = s->pts;
s->pts += window_size;
for (ch = 0; ch < inlink->channels; ch++) {
dst = (float *)out->extended_data[ch];
buf = (float *)s->buffer->extended_data[ch];
for (n = 0; n < window_size; n++) {
dst[n] = buf[n] * (1 - s->overlap);
}
memmove(buf, buf + window_size, window_size * 4);
}
ret = ff_filter_frame(outlink, out);
if (ret < 0)
break;
}
av_audio_fifo_drain(s->fifo, s->hop_size);
}
av_frame_free(&in);
return ret;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AFFTFiltContext *s = ctx->priv;
int i;
av_fft_end(s->fft);
av_fft_end(s->ifft);
for (i = 0; i < s->nb_exprs; i++) {
if (s->fft_data)
av_freep(&s->fft_data[i]);
}
av_freep(&s->fft_data);
for (i = 0; i < s->nb_exprs; i++) {
av_expr_free(s->real[i]);
av_expr_free(s->imag[i]);
}
av_freep(&s->real);
av_freep(&s->imag);
av_frame_free(&s->buffer);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_afftfilt = {
.name = "afftfilt",
.description = NULL_IF_CONFIG_SMALL("Apply arbitrary expressions to samples in frequency domain."),
.priv_size = sizeof(AFFTFiltContext),
.priv_class = &afftfilt_class,
.inputs = inputs,
.outputs = outputs,
.query_formats = query_formats,
.uninit = uninit,
};
libavfilter/allfilters.c
View file @ fa04ec72
......@@ -52,6 +52,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(AEMPHASIS, aemphasis, af);
REGISTER_FILTER(AEVAL, aeval, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFFTFILT, afftfilt, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AGATE, agate, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
......
libavfilter/version.h
View file @ fa04ec72
......@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 25
#define LIBAVFILTER_VERSION_MINOR 26
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
......
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