[FFmpeg-devel] [PATCH] avfilter: add afftfilter
Paul B Mahol
onemda at gmail.com
Mon Jan 18 19:35:28 CET 2016
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
configure | 3 +
doc/filters.texi | 77 +++++++++
libavfilter/Makefile | 1 +
libavfilter/af_afftfilt.c | 396 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
5 files changed, 478 insertions(+)
create mode 100644 libavfilter/af_afftfilt.c
diff --git a/configure b/configure
index cdf07ae..b74ea9f 100755
--- a/configure
+++ b/configure
@@ -2840,6 +2840,8 @@ 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"
@@ -6063,6 +6065,7 @@ 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"
diff --git a/doc/filters.texi b/doc/filters.texi
index d8e3317..1a03995 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -733,6 +733,83 @@ afade=t=out:st=875:d=25
@end example
@end itemize
+ at section afftfilt
+Apply arbitrary expressions to samples in frequency domain
+
+ at table @option
+ at 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.
+
+ at 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:
+
+ at table @option
+ at item sr
+sample rate
+
+ at item b
+current frequency bin number
+
+ at item nb
+number of available bins
+
+ at item ch
+channel number of the current expression
+
+ at item chs
+number of channels
+
+ at item pts
+current frame pts
+ at end table
+
+ at item win_size
+Set window size.
+
+It accepts the following values:
+ at table @samp
+ at item w16
+ at item w32
+ at item w64
+ at item w128
+ at item w256
+ at item w512
+ at item w1024
+ at item w2048
+ at item w4096
+ at item w8192
+ at item w16384
+ at item w32768
+ at item w65536
+ at end table
+Default is @code{w4096}
+
+ at item win_func
+Set window function. Default is @code{hann}.
+
+ at item overlap
+Set window overlap. If set to 1, the recommended overlap for selected
+window function will be picked. Default is @code{0.75}.
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+Increase first 50 bins by 0.1 and lower all other frequencies by factor of 10:
+ at example
+afftfilt="1.1*between(b\,0\,50)+0.1*between(b\,50\,f)"
+ at end example
+ at end itemize
+
@anchor{aformat}
@section aformat
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e3e3561..242f56d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_afftfilt.c b/libavfilter/af_afftfilt.c
new file mode 100644
index 0000000..9643627
--- /dev/null
+++ b/libavfilter/af_afftfilt.c
@@ -0,0 +1,396 @@
+/*
+ * 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 *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;
+ 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;
+
+ 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. / window_size;
+ 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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1faf393..f270bdf 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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);
--
1.9.1
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