[FFmpeg-devel] [PATCH] avfilter: add arbitrary audio FIR filter
Muhammad Faiz
mfcc64 at gmail.com
Mon May 8 09:01:50 EEST 2017
On Mon, May 8, 2017 at 1:22 AM, Paul B Mahol <onemda at gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> configure | 2 +
> doc/filters.texi | 30 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_afir.c | 541 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 5 files changed, 575 insertions(+)
> create mode 100644 libavfilter/af_afir.c
>
> diff --git a/configure b/configure
> index b3cb5b0..0d83c6a 100755
> --- a/configure
> +++ b/configure
> @@ -3078,6 +3078,8 @@ unix_protocol_select="network"
> # filters
> afftfilt_filter_deps="avcodec"
> afftfilt_filter_select="fft"
> +afir_filter_deps="avcodec"
> +afir_filter_select="fft"
> amovie_filter_deps="avcodec avformat"
> aresample_filter_deps="swresample"
> ass_filter_deps="libass"
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 119e747..7b6a67b 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -878,6 +878,36 @@ afftfilt="1-clip((b/nb)*b,0,1)"
> @end example
> @end itemize
>
> + at section afir
> +
> +Apply an Arbitary Frequency Impulse Response filter.
> +
> +This filter uses second stream as FIR coefficients.
> +If second stream holds single channel, it will be used
> +for all input channels in first stream, otherwise
> +number of channels in second stream must be same as
> +number of channels in first stream.
> +
> +It accepts the following parameters:
> +
> + at table @option
> + at item dry
> +Set dry gain. This sets input gain.
> +
> + at item wet
> +Set wet gain. This sets final output gain.
> +
> + at item envelope
> +How much to fade out Impulse Response to the end.
> +
> + at item length
> +Set Impulse Response filter length. Default is 1, which means whole IR is processed.
> +
> + at item auto
> +Enable auto gain calculation of Impulse Response coefficients.
> +By default is enabled.
> + at end table
> +
Probably, these options aren't required if algo is correct.
> @anchor{aformat}
> @section aformat
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 66c36e4..c797eb5 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -38,6 +38,7 @@ OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
> OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
> OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o window_func.o
> +OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
> OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
> OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
> diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
> new file mode 100644
> index 0000000..7f0afce
> --- /dev/null
> +++ b/libavfilter/af_afir.c
> @@ -0,0 +1,541 @@
> +/*
> + * Copyright (c) 2017 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
> + */
> +
> +/**
> + * @file
> + * An arbitrary audio FIR filter
> + */
> +
> +#include "libavutil/audio_fifo.h"
> +#include "libavutil/common.h"
> +#include "libavutil/opt.h"
> +#include "libavcodec/avfft.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "formats.h"
> +#include "internal.h"
> +
> +#define MAX_IR_DURATION 30
> +
> +typedef struct AudioFIRContext {
> + const AVClass *class;
> +
> + float wet_gain;
> + float dry_gain;
> + float envelope;
> + float length;
> + int auto_gain;
> +
> + float gain;
> +
> + int eof_coeffs;
> + int have_coeffs;
> + int nb_coeffs;
> + int nb_taps;
> + int part_size;
> + int nb_partitions;
> + int nb_channels;
> + int ir_length;
> + int fft_length;
> + int nb_coef_channels;
> + int one2many;
> + int nb_samples;
> + int want_skip;
> + int need_padding;
> +
> + RDFTContext **rdft, **irdft;
> + float **sum;
> + float **block;
> + FFTComplex **coeff;
> +
> + AVAudioFifo *fifo[2];
> + AVFrame *in[2];
> + AVFrame *buffer;
> + int64_t pts;
> + int index;
> +} AudioFIRContext;
> +
> +static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
> +{
> + AudioFIRContext *s = ctx->priv;
> + const FFTComplex *coeff = s->coeff[ch * !s->one2many];
> + const float *src = (const float *)s->in[0]->extended_data[ch];
> + int index1 = (s->index + 1) % 3;
> + int index2 = (s->index + 2) % 3;
> + float *block = s->block[ch];
> + float *sum = s->sum[ch];
> + AVFrame *out = arg;
> + float *dst;
> + int n, i;
> +
> + memset(sum, 0, sizeof(*sum) * s->fft_length);
> + memset(block, 0, sizeof(*block) * s->fft_length);
> + for (n = 0; n < s->nb_samples; n++) {
> + block[s->part_size + n] = src[n] * s->dry_gain;
> + }
> +
> + av_rdft_calc(s->rdft[ch], block);
> + block[2 * s->part_size] = block[1];
> + block[1] = 0;
> +
> + for (i = 0; i < s->nb_partitions; i++) {
> + const int coffset = i * (s->part_size + 1);
> +
> + for (n = 0; n < s->part_size; n++) {
> + const float cre = coeff[coffset + n].re;
> + const float cim = coeff[coffset + n].im;
> + const float tre = block[2 * n ];
> + const float tim = block[2 * n + 1];
> +
> + sum[2 * n ] += tre * cre - tim * cim;
> + sum[2 * n + 1] += tre * cim + tim * cre;
> + }
> + sum[2 * n] += block[2 * n] * coeff[coffset + n].re;
> + }
This is still wrong.
As I read in articles, each ir partition is convoluted with different
data block.
Test:
aevalsrc = 'if(n, 0, 1)',
firequalizer =
delay = 0.023:
fixed = on:
wfunc = nuttall:
gain = 'if(between(f, 1000, 5000), -INF, 0)',
atrim = end_sample = 2048 [ir];
aevalsrc = '0.7*sin(3000*t*t)' [data];
[data][ir]
afir,
asplit [out0],
showspectrum=s=1024x512:win_func=nuttall,
format = rgb24 [out1]
> +
> + sum[1] = sum[2 * n];
> + av_rdft_calc(s->irdft[ch], sum);
> +
> + dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
> + for (n = 0; n < s->part_size; n++) {
> + dst[n] += sum[n];
> + }
> +
> + dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
> +
> + memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
> +
> + dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
> +
> + if (out) {
> + float *ptr = (float *)out->extended_data[ch];
> + for (n = 0; n < out->nb_samples; n++) {
> + ptr[n] = dst[n] * s->gain * s->wet_gain;
> + }
> + }
> +
> + return 0;
> +}
> +
> +static int fir_frame(AudioFIRContext *s, AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + AVFrame *out = NULL;
> + int ret;
> +
> + s->nb_samples = FFMIN(s->part_size, av_audio_fifo_size(s->fifo[0]));
> +
> + if (!s->want_skip) {
> + out = ff_get_audio_buffer(outlink, s->nb_samples);
> + if (!out)
> + return AVERROR(ENOMEM);
> + }
> +
> + s->in[0] = ff_get_audio_buffer(ctx->inputs[0], s->nb_samples);
> + if (!s->in[0]) {
> + av_frame_free(&out);
> + return AVERROR(ENOMEM);
> + }
> +
> + av_audio_fifo_peek(s->fifo[0], (void **)s->in[0]->extended_data, s->nb_samples);
> +
> + ctx->internal->execute(ctx, fir_channel, out, NULL, outlink->channels);
> +
> + av_audio_fifo_drain(s->fifo[0], s->nb_samples);
> +
> + if (!s->want_skip) {
> + out->pts = s->pts;
> + if (s->pts != AV_NOPTS_VALUE)
> + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
> + }
> +
> + s->index++;
> + if (s->index == 3)
> + s->index = 0;
> +
> + av_frame_free(&s->in[0]);
> +
> + if (s->want_skip == 1) {
> + s->want_skip = 0;
> + ret = 0;
> + } else {
> + ret = ff_filter_frame(outlink, out);
> + }
> +
> + return ret;
> +}
> +
> +static int convert_coeffs(AVFilterContext *ctx)
> +{
> + AudioFIRContext *s = ctx->priv;
> + int i, ch, n, N;
> + float power = 0;
> +
> + s->nb_taps = av_audio_fifo_size(s->fifo[1]);
> +
> + for (n = 4; (1 << n) < s->nb_taps; n++);
> + N = FFMIN(n, 16);
> + s->ir_length = 1 << n;
> + s->fft_length = (1 << (N + 1)) + 1;
> + s->part_size = 1 << (N - 1);
> + s->nb_partitions = (s->nb_taps + s->part_size - 1) / s->part_size;
> + s->nb_coeffs = s->ir_length + s->nb_partitions;
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->sum[ch] = av_calloc(s->fft_length, sizeof(**s->sum));
> + if (!s->sum[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + s->coeff[ch] = av_calloc(s->nb_coeffs, sizeof(**s->coeff));
> + if (!s->coeff[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->block[ch] = av_calloc(s->fft_length, sizeof(**s->block));
> + if (!s->block[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < ctx->inputs[0]->channels; ch++) {
> + s->rdft[ch] = av_rdft_init(N, DFT_R2C);
> + s->irdft[ch] = av_rdft_init(N, IDFT_C2R);
> + if (!s->rdft[ch] || !s->irdft[ch])
> + return AVERROR(ENOMEM);
> + }
> +
> + s->in[1] = ff_get_audio_buffer(ctx->inputs[1], s->nb_taps);
> + if (!s->in[1])
> + return AVERROR(ENOMEM);
> +
> + s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
> + if (!s->buffer)
> + return AVERROR(ENOMEM);
> +
> + av_audio_fifo_read(s->fifo[1], (void **)s->in[1]->extended_data, s->nb_taps);
> +
> + for (ch = 0; ch < ctx->inputs[1]->channels; ch++) {
> + float *time = (float *)s->in[1]->extended_data[!s->one2many * ch];
> + float *block = s->block[ch];
> + FFTComplex *coeff = s->coeff[ch];
> +
> + for (i = 0; i < FFMAX(1, s->length * s->nb_taps); i++) {
> + float gain = s->envelope + (1 - s->envelope) * exp(-1.0 * (double)i / s->nb_taps / .25);
> + time[i] *= gain;
> + }
> +
> + for (; i < s->nb_taps; i++)
> + time[i] = 0;
> +
> + for (i = 0; i < s->nb_partitions; i++) {
> + const float scale = 1.f / s->part_size;
> + const int toffset = i * s->part_size;
> + const int coffset = i * (s->part_size + 1);
> + const int boffset = s->part_size;
> + const int remaining = s->nb_taps - (i * s->part_size);
> + const int size = remaining >= s->part_size ? s->part_size : remaining;
> +
> + memset(block, 0, sizeof(*block) * s->fft_length);
> + for (n = 0; n < size; n++) {
> + power += time[n + toffset] * time[n + toffset];
> + block[n + boffset] = time[n + toffset];
> + }
> +
> + av_rdft_calc(s->rdft[0], block);
> +
> + coeff[coffset].re = block[0] * scale;
> + coeff[coffset].im = 0;
> + for (n = 1; n < s->part_size; n++) {
> + coeff[coffset + n].re = block[2 * n] * scale;
> + coeff[coffset + n].im = block[2 * n + 1] * scale;
> + }
> + coeff[coffset + s->part_size].re = block[1] * scale;
> + coeff[coffset + s->part_size].im = 0;
> + }
> + }
> + power /= ctx->inputs[1]->channels;
> +
> + av_frame_free(&s->in[1]);
> + s->gain = .5f / (s->auto_gain ? sqrtf(power) : sqrtf(s->part_size));
> + av_log(ctx, AV_LOG_DEBUG, "nb_taps: %d\n", s->nb_taps);
> + av_log(ctx, AV_LOG_DEBUG, "nb_partitions: %d\n", s->nb_partitions);
> + av_log(ctx, AV_LOG_DEBUG, "partition size: %d\n", s->part_size);
> + av_log(ctx, AV_LOG_DEBUG, "ir_length: %d\n", s->ir_length);
> +
> + s->have_coeffs = 1;
> +
> + return 0;
> +}
> +
> +static int read_ir(AVFilterLink *link, AVFrame *frame)
> +{
> + AVFilterContext *ctx = link->dst;
> + AudioFIRContext *s = ctx->priv;
> + int nb_taps, max_nb_taps;
> +
> + av_audio_fifo_write(s->fifo[1], (void **)frame->extended_data,
> + frame->nb_samples);
> + av_frame_free(&frame);
> +
> + nb_taps = av_audio_fifo_size(s->fifo[1]);
> + max_nb_taps = MAX_IR_DURATION * ctx->outputs[0]->sample_rate;
> + if (nb_taps > max_nb_taps) {
> + av_log(ctx, AV_LOG_ERROR, "Too big number of coefficients: %d > %d.\n", nb_taps, max_nb_taps);
> + return AVERROR(EINVAL);
> + }
> +
> + return 0;
> +}
> +
> +static int filter_frame(AVFilterLink *link, AVFrame *frame)
> +{
> + AVFilterContext *ctx = link->dst;
> + AudioFIRContext *s = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> + int ret = 0;
> +
> + av_audio_fifo_write(s->fifo[0], (void **)frame->extended_data,
> + frame->nb_samples);
> + if (s->pts == AV_NOPTS_VALUE)
> + s->pts = frame->pts;
> +
> + av_frame_free(&frame);
> +
> + if (!s->have_coeffs && s->eof_coeffs) {
> + ret = convert_coeffs(ctx);
> + if (ret < 0)
> + return ret;
> + }
> +
> + if (s->have_coeffs) {
> + while (av_audio_fifo_size(s->fifo[0]) >= s->part_size) {
> + ret = fir_frame(s, outlink);
> + if (ret < 0)
> + break;
> + }
> + }
> + return ret;
> +}
> +
> +static int request_frame(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + AudioFIRContext *s = ctx->priv;
> + int ret;
> +
> + if (!s->eof_coeffs) {
> + ret = ff_request_frame(ctx->inputs[1]);
> + if (ret == AVERROR_EOF) {
> + s->eof_coeffs = 1;
> + ret = 0;
> + }
> + return ret;
> + }
> + ret = ff_request_frame(ctx->inputs[0]);
> + if (ret == AVERROR_EOF && s->have_coeffs) {
> + if (s->need_padding) {
> + AVFrame *silence = ff_get_audio_buffer(outlink, s->part_size);
> +
> + if (!silence)
> + return AVERROR(ENOMEM);
> + av_audio_fifo_write(s->fifo[0], (void **)silence->extended_data,
> + silence->nb_samples);
> + av_frame_free(&silence);
> + s->need_padding = 0;
> + }
> +
> + while (av_audio_fifo_size(s->fifo[0]) > 0) {
> + ret = fir_frame(s, outlink);
> + if (ret < 0)
> + return ret;
> + }
> + ret = AVERROR_EOF;
> + }
> + 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, i;
> +
> + layouts = ff_all_channel_counts();
> + if ((ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
> + return ret;
> +
> + for (i = 0; i < 2; i++) {
> + layouts = ff_all_channel_counts();
> + if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
> + return ret;
> + }
> +
> + formats = ff_make_format_list(sample_fmts);
> + if ((ret = ff_set_common_formats(ctx, formats)) < 0)
> + return ret;
> +
> + formats = ff_all_samplerates();
> + return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + AudioFIRContext *s = ctx->priv;
> +
> + if (ctx->inputs[0]->channels != ctx->inputs[1]->channels &&
> + ctx->inputs[1]->channels != 1) {
> + av_log(ctx, AV_LOG_ERROR,
> + "Second input must have same number of channels as first input or "
> + "exactly 1 channel.\n");
> + return AVERROR(EINVAL);
> + }
> +
> + s->one2many = ctx->inputs[1]->channels == 1;
> + outlink->sample_rate = ctx->inputs[0]->sample_rate;
> + outlink->time_base = ctx->inputs[0]->time_base;
> + outlink->channel_layout = ctx->inputs[0]->channel_layout;
> + outlink->channels = ctx->inputs[0]->channels;
> +
> + s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
> + s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
> + if (!s->fifo[0] || !s->fifo[1])
> + return AVERROR(ENOMEM);
> +
> + s->sum = av_calloc(outlink->channels, sizeof(*s->sum));
> + s->coeff = av_calloc(ctx->inputs[1]->channels, sizeof(*s->coeff));
> + s->block = av_calloc(ctx->inputs[0]->channels, sizeof(*s->block));
> + s->rdft = av_calloc(outlink->channels, sizeof(*s->rdft));
> + s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
> + if (!s->sum || !s->coeff || !s->block || !s->rdft || !s->irdft)
> + return AVERROR(ENOMEM);
> +
> + s->nb_channels = outlink->channels;
> + s->nb_coef_channels = ctx->inputs[1]->channels;
> + s->want_skip = 1;
> + s->need_padding = 1;
> + s->pts = AV_NOPTS_VALUE;
> +
> + return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + AudioFIRContext *s = ctx->priv;
> + int ch;
> +
> + if (s->sum) {
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + av_freep(&s->sum[ch]);
> + }
> + }
> + av_freep(&s->sum);
> +
> + if (s->coeff) {
> + for (ch = 0; ch < s->nb_coef_channels; ch++) {
> + av_freep(&s->coeff[ch]);
> + }
> + }
> + av_freep(&s->coeff);
> +
> + if (s->block) {
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + av_freep(&s->block[ch]);
> + }
> + }
> + av_freep(&s->block);
> +
> + if (s->rdft) {
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + av_rdft_end(s->rdft[ch]);
> + }
> + }
> + av_freep(&s->rdft);
> +
> + if (s->irdft) {
> + for (ch = 0; ch < s->nb_channels; ch++) {
> + av_rdft_end(s->irdft[ch]);
> + }
> + }
> + av_freep(&s->irdft);
> +
> + av_frame_free(&s->in[0]);
> + av_frame_free(&s->in[1]);
> + av_frame_free(&s->buffer);
> +
> + av_audio_fifo_free(s->fifo[0]);
> + av_audio_fifo_free(s->fifo[1]);
> +}
> +
> +static const AVFilterPad afir_inputs[] = {
> + {
> + .name = "main",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + },{
> + .name = "ir",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = read_ir,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad afir_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_output,
> + .request_frame = request_frame,
> + },
> + { NULL }
> +};
> +
> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +#define OFFSET(x) offsetof(AudioFIRContext, x)
> +
> +static const AVOption afir_options[] = {
> + { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> + { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> + { "envelope", "set IR envelope", OFFSET(envelope), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> + { "length", "set IR length", OFFSET(length), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
> + { "auto", "enable auto-gain", OFFSET(auto_gain), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(afir);
> +
> +AVFilter ff_af_afir = {
> + .name = "afir",
> + .description = NULL_IF_CONFIG_SMALL("Apply Finite Impulse Response filter with supplied coefficients in 2nd stream."),
> + .priv_size = sizeof(AudioFIRContext),
> + .priv_class = &afir_class,
> + .query_formats = query_formats,
> + .uninit = uninit,
> + .inputs = afir_inputs,
> + .outputs = afir_outputs,
> + .flags = AVFILTER_FLAG_SLICE_THREADS,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 8fb87eb..555c442 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -50,6 +50,7 @@ static void register_all(void)
> REGISTER_FILTER(AEVAL, aeval, af);
> REGISTER_FILTER(AFADE, afade, af);
> REGISTER_FILTER(AFFTFILT, afftfilt, af);
> + REGISTER_FILTER(AFIR, afir, af);
> REGISTER_FILTER(AFORMAT, aformat, af);
> REGISTER_FILTER(AGATE, agate, af);
> REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
> --
> 2.9.3
>
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