avfilter: add firequalizer filter

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>

8 files changed, 708 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 7d672fbdf6..fa4400c6ce 100644
--- a/Changelog
+++ b/Changelog
@@ -6,6 +6,7 @@ version <next>:
 - fieldhint filter
 - loop video filter and aloop audio filter
 - Bob Weaver deinterlacing filter
+- firequalizer filter
 
 
 version 3.0:
diff --git a/MAINTAINERS b/MAINTAINERS
index 0705a6999f..f518aed78b 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -353,6 +353,7 @@ Filters:
   af_biquads.c                          Paul B Mahol
   af_chorus.c                           Paul B Mahol
   af_compand.c                          Paul B Mahol
+  af_firequalizer.c                     Muhammad Faiz
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
   af_sidechaincompress.c                Paul B Mahol
diff --git a/configure b/configure
index 6b3ee5fd86..3d1ee49062 100755
--- a/configure
+++ b/configure
@@ -2861,6 +2861,8 @@ eq_filter_deps="gpl"
 fftfilt_filter_deps="avcodec"
 fftfilt_filter_select="rdft"
 find_rect_filter_deps="avcodec avformat gpl"
+firequalizer_filter_deps="avcodec"
+firequalizer_filter_select="rdft"
 flite_filter_deps="libflite"
 frei0r_filter_deps="frei0r dlopen"
 frei0r_src_filter_deps="frei0r dlopen"
diff --git a/doc/filters.texi b/doc/filters.texi
index 250367e85e..aca76630c0 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2366,6 +2366,115 @@ Sets the difference coefficient (default: 2.5). 0.0 means mono sound
 Enable clipping. By default is enabled.
 @end table
 
+@section firequalizer
+Apply FIR Equalization using arbitrary frequency response.
+
+The filter accepts the following option:
+
+@table @option
+@item gain
+Set gain curve equation (in dB). The expression can contain variables:
+@table @option
+@item f
+the evaluated frequency
+@item sr
+sample rate
+@item ch
+channel number, set to 0 when multichannels evaluation is disabled
+@item chid
+channel id, see libavutil/channel_layout.h, set to the first channel id when
+multichannels evaluation is disabled
+@item chs
+number of channels
+@item chlayout
+channel_layout, see libavutil/channel_layout.h
+
+@end table
+and functions:
+@table @option
+@item gain_interpolate(f)
+interpolate gain on frequency f based on gain_entry
+@end table
+This option is also available as command. Default is @code{gain_interpolate(f)}.
+
+@item gain_entry
+Set gain entry for gain_interpolate function. The expression can
+contain functions:
+@table @option
+@item entry(f, g)
+store gain entry at frequency f with value g
+@end table
+This option is also available as command.
+
+@item delay
+Set filter delay in seconds. Higher value means more accurate.
+Default is @code{0.01}.
+
+@item accuracy
+Set filter accuracy in Hz. Lower value means more accurate.
+Default is @code{5}.
+
+@item wfunc
+Set window function. Acceptable values are:
+@table @option
+@item rectangular
+rectangular window, useful when gain curve is already smooth
+@item hann
+hann window (default)
+@item hamming
+hamming window
+@item blackman
+blackman window
+@item nuttall3
+3-terms continuous 1st derivative nuttall window
+@item mnuttall3
+minimum 3-terms discontinuous nuttall window
+@item nuttall
+4-terms continuous 1st derivative nuttall window
+@item bnuttall
+minimum 4-terms discontinuous nuttall (blackman-nuttall) window
+@item bharris
+blackman-harris window
+@end table
+
+@item fixed
+If enabled, use fixed number of audio samples. This improves speed when
+filtering with large delay. Default is disabled.
+
+@item multi
+Enable multichannels evaluation on gain. Default is disabled.
+@end table
+
+@subsection Examples
+@itemize
+@item
+lowpass at 1000 Hz:
+@example
+firequalizer=gain='if(lt(f,1000), 0, -INF)'
+@end example
+@item
+lowpass at 1000 Hz with gain_entry:
+@example
+firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
+@end example
+@item
+custom equalization:
+@example
+firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
+@end example
+@item
+higher delay:
+@example
+firequalizer=delay=0.1:fixed=on
+@end example
+@item
+lowpass on left channel, highpass on right channel:
+@example
+firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
+:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
+@end example
+@end itemize
+
 @section flanger
 Apply a flanging effect to the audio.
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 10c2e0bdb7..a8469fd224 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -80,6 +80,7 @@ OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
 OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
 OBJS-$(CONFIG_EXTRASTEREO_FILTER)            += af_extrastereo.o
+OBJS-$(CONFIG_FIREQUALIZER_FILTER)           += af_firequalizer.o
 OBJS-$(CONFIG_FLANGER_FILTER)                += af_flanger.o generate_wave_table.o
 OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o
 OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o
diff --git a/libavfilter/af_firequalizer.c b/libavfilter/af_firequalizer.c
new file mode 100644
index 0000000000..4c9d95e495
--- /dev/null
+++ b/libavfilter/af_firequalizer.c
@@ -0,0 +1,592 @@
+/*
+ * Copyright (c) 2016 Muhammad Faiz <mfcc64@gmail.com>
+ *
+ * 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/opt.h"
+#include "libavutil/eval.h"
+#include "libavutil/avassert.h"
+#include "libavcodec/avfft.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+#define RDFT_BITS_MIN 4
+#define RDFT_BITS_MAX 16
+
+enum WindowFunc {
+    WFUNC_MIN,
+    WFUNC_RECTANGULAR = WFUNC_MIN,
+    WFUNC_HANN,
+    WFUNC_HAMMING,
+    WFUNC_BLACKMAN,
+    WFUNC_NUTTALL3,
+    WFUNC_MNUTTALL3,
+    WFUNC_NUTTALL,
+    WFUNC_BNUTTALL,
+    WFUNC_BHARRIS,
+    WFUNC_MAX = WFUNC_BHARRIS
+};
+
+#define NB_GAIN_ENTRY_MAX 4096
+typedef struct {
+    double  freq;
+    double  gain;
+} GainEntry;
+
+typedef struct {
+    int buf_idx;
+    int overlap_idx;
+} OverlapIndex;
+
+typedef struct {
+    const AVClass *class;
+
+    RDFTContext   *analysis_irdft;
+    RDFTContext   *rdft;
+    RDFTContext   *irdft;
+    int           analysis_rdft_len;
+    int           rdft_len;
+
+    float         *analysis_buf;
+    float         *kernel_tmp_buf;
+    float         *kernel_buf;
+    float         *conv_buf;
+    OverlapIndex  *conv_idx;
+    int           fir_len;
+    int           nsamples_max;
+    int64_t       next_pts;
+    int           frame_nsamples_max;
+    int           remaining;
+
+    char          *gain_cmd;
+    char          *gain_entry_cmd;
+    const char    *gain;
+    const char    *gain_entry;
+    double        delay;
+    double        accuracy;
+    int           wfunc;
+    int           fixed;
+    int           multi;
+
+    int           nb_gain_entry;
+    int           gain_entry_err;
+    GainEntry     gain_entry_tbl[NB_GAIN_ENTRY_MAX];
+} FIREqualizerContext;
+
+#define OFFSET(x) offsetof(FIREqualizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption firequalizer_options[] = {
+    { "gain", "set gain curve", OFFSET(gain), AV_OPT_TYPE_STRING, { .str = "gain_interpolate(f)" }, 0, 0, FLAGS },
+    { "gain_entry", "set gain entry", OFFSET(gain_entry), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, FLAGS },
+    { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.0, 1e10, FLAGS },
+    { "accuracy", "set accuracy", OFFSET(accuracy), AV_OPT_TYPE_DOUBLE, { .dbl = 5.0 }, 0.0, 1e10, FLAGS },
+    { "wfunc", "set window function", OFFSET(wfunc), AV_OPT_TYPE_INT, { .i64 = WFUNC_HANN }, WFUNC_MIN, WFUNC_MAX, FLAGS, "wfunc" },
+        { "rectangular", "rectangular window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_RECTANGULAR }, 0, 0, FLAGS, "wfunc" },
+        { "hann", "hann window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HANN }, 0, 0, FLAGS, "wfunc" },
+        { "hamming", "hamming window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HAMMING }, 0, 0, FLAGS, "wfunc" },
+        { "blackman", "blackman window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BLACKMAN }, 0, 0, FLAGS, "wfunc" },
+        { "nuttall3", "3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL3 }, 0, 0, FLAGS, "wfunc" },
+        { "mnuttall3", "minimum 3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_MNUTTALL3 }, 0, 0, FLAGS, "wfunc" },
+        { "nuttall", "nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL }, 0, 0, FLAGS, "wfunc" },
+        { "bnuttall", "blackman-nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BNUTTALL }, 0, 0, FLAGS, "wfunc" },
+        { "bharris", "blackman-harris window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BHARRIS }, 0, 0, FLAGS, "wfunc" },
+    { "fixed", "set fixed frame samples", OFFSET(fixed), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
+    { "multi", "set multi channels mode", OFFSET(multi), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(firequalizer);
+
+static void common_uninit(FIREqualizerContext *s)
+{
+    av_rdft_end(s->analysis_irdft);
+    av_rdft_end(s->rdft);
+    av_rdft_end(s->irdft);
+    s->analysis_irdft = s->rdft = s->irdft = NULL;
+
+    av_freep(&s->analysis_buf);
+    av_freep(&s->kernel_tmp_buf);
+    av_freep(&s->kernel_buf);
+    av_freep(&s->conv_buf);
+    av_freep(&s->conv_idx);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    FIREqualizerContext *s = ctx->priv;
+
+    common_uninit(s);
+    av_freep(&s->gain_cmd);
+    av_freep(&s->gain_entry_cmd);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    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 void fast_convolute(FIREqualizerContext *s, const float *kernel_buf, float *conv_buf,
+                           OverlapIndex *idx, float *data, int nsamples)
+{
+    if (nsamples <= s->nsamples_max) {
+        float *buf = conv_buf + idx->buf_idx * s->rdft_len;
+        float *obuf = conv_buf + !idx->buf_idx * s->rdft_len + idx->overlap_idx;
+        int k;
+
+        memcpy(buf, data, nsamples * sizeof(*data));
+        memset(buf + nsamples, 0, (s->rdft_len - nsamples) * sizeof(*data));
+        av_rdft_calc(s->rdft, buf);
+
+        buf[0] *= kernel_buf[0];
+        buf[1] *= kernel_buf[1];
+        for (k = 2; k < s->rdft_len; k += 2) {
+            float re, im;
+            re = buf[k] * kernel_buf[k] - buf[k+1] * kernel_buf[k+1];
+            im = buf[k] * kernel_buf[k+1] + buf[k+1] * kernel_buf[k];
+            buf[k] = re;
+            buf[k+1] = im;
+        }
+
+        av_rdft_calc(s->irdft, buf);
+        for (k = 0; k < s->rdft_len - idx->overlap_idx; k++)
+            buf[k] += obuf[k];
+        memcpy(data, buf, nsamples * sizeof(*data));
+        idx->buf_idx = !idx->buf_idx;
+        idx->overlap_idx = nsamples;
+    } else {
+        while (nsamples > s->nsamples_max * 2) {
+            fast_convolute(s, kernel_buf, conv_buf, idx, data, s->nsamples_max);
+            data += s->nsamples_max;
+            nsamples -= s->nsamples_max;
+        }
+        fast_convolute(s, kernel_buf, conv_buf, idx, data, nsamples/2);
+        fast_convolute(s, kernel_buf, conv_buf, idx, data + nsamples/2, nsamples - nsamples/2);
+    }
+}
+
+static double entry_func(void *p, double freq, double gain)
+{
+    AVFilterContext *ctx = p;
+    FIREqualizerContext *s = ctx->priv;
+
+    if (s->nb_gain_entry >= NB_GAIN_ENTRY_MAX) {
+        av_log(ctx, AV_LOG_ERROR, "entry table overflow.\n");
+        s->gain_entry_err = AVERROR(EINVAL);
+        return 0;
+    }
+
+    if (isnan(freq)) {
+        av_log(ctx, AV_LOG_ERROR, "nan frequency (%g, %g).\n", freq, gain);
+        s->gain_entry_err = AVERROR(EINVAL);
+        return 0;
+    }
+
+    if (s->nb_gain_entry > 0 && freq <= s->gain_entry_tbl[s->nb_gain_entry - 1].freq) {
+        av_log(ctx, AV_LOG_ERROR, "unsorted frequency (%g, %g).\n", freq, gain);
+        s->gain_entry_err = AVERROR(EINVAL);
+        return 0;
+    }
+
+    s->gain_entry_tbl[s->nb_gain_entry].freq = freq;
+    s->gain_entry_tbl[s->nb_gain_entry].gain = gain;
+    s->nb_gain_entry++;
+    return 0;
+}
+
+static int gain_entry_compare(const void *key, const void *memb)
+{
+    const double *freq = key;
+    const GainEntry *entry = memb;
+
+    if (*freq < entry[0].freq)
+        return -1;
+    if (*freq > entry[1].freq)
+        return 1;
+    return 0;
+}
+
+static double gain_interpolate_func(void *p, double freq)
+{
+    AVFilterContext *ctx = p;
+    FIREqualizerContext *s = ctx->priv;
+    GainEntry *res;
+    double d0, d1, d;
+
+    if (isnan(freq))
+        return freq;
+
+    if (!s->nb_gain_entry)
+        return 0;
+
+    if (freq <= s->gain_entry_tbl[0].freq)
+        return s->gain_entry_tbl[0].gain;
+
+    if (freq >= s->gain_entry_tbl[s->nb_gain_entry-1].freq)
+        return s->gain_entry_tbl[s->nb_gain_entry-1].gain;
+
+    res = bsearch(&freq, &s->gain_entry_tbl, s->nb_gain_entry - 1, sizeof(*res), gain_entry_compare);
+    av_assert0(res);
+
+    d  = res[1].freq - res[0].freq;
+    d0 = freq - res[0].freq;
+    d1 = res[1].freq - freq;
+
+    if (d0 && d1)
+        return (d0 * res[1].gain + d1 * res[0].gain) / d;
+
+    if (d0)
+        return res[1].gain;
+
+    return res[0].gain;
+}
+
+static const char *const var_names[] = {
+    "f",
+    "sr",
+    "ch",
+    "chid",
+    "chs",
+    "chlayout",
+    NULL
+};
+
+enum VarOffset {
+    VAR_F,
+    VAR_SR,
+    VAR_CH,
+    VAR_CHID,
+    VAR_CHS,
+    VAR_CHLAYOUT,
+    VAR_NB
+};
+
+static int generate_kernel(AVFilterContext *ctx, const char *gain, const char *gain_entry)
+{
+    FIREqualizerContext *s = ctx->priv;
+    AVFilterLink *inlink = ctx->inputs[0];
+    const char *gain_entry_func_names[] = { "entry", NULL };
+    const char *gain_func_names[] = { "gain_interpolate", NULL };
+    double (*gain_entry_funcs[])(void *, double, double) = { entry_func, NULL };
+    double (*gain_funcs[])(void *, double) = { gain_interpolate_func, NULL };
+    double vars[VAR_NB];
+    AVExpr *gain_expr;
+    int ret, k, center, ch;
+
+    s->nb_gain_entry = 0;
+    s->gain_entry_err = 0;
+    if (gain_entry) {
+        double result = 0.0;
+        ret = av_expr_parse_and_eval(&result, gain_entry, NULL, NULL, NULL, NULL,
+                                     gain_entry_func_names, gain_entry_funcs, ctx, 0, ctx);
+        if (ret < 0)
+            return ret;
+        if (s->gain_entry_err < 0)
+            return s->gain_entry_err;
+    }
+
+    av_log(ctx, AV_LOG_DEBUG, "nb_gain_entry = %d.\n", s->nb_gain_entry);
+
+    ret = av_expr_parse(&gain_expr, gain, var_names,
+                        gain_func_names, gain_funcs, NULL, NULL, 0, ctx);
+    if (ret < 0)
+        return ret;
+
+    vars[VAR_CHS] = inlink->channels;
+    vars[VAR_CHLAYOUT] = inlink->channel_layout;
+    vars[VAR_SR] = inlink->sample_rate;
+    for (ch = 0; ch < inlink->channels; ch++) {
+        vars[VAR_CH] = ch;
+        vars[VAR_CHID] = av_channel_layout_extract_channel(inlink->channel_layout, ch);
+        vars[VAR_F] = 0.0;
+        s->analysis_buf[0] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+        vars[VAR_F] = 0.5 * inlink->sample_rate;
+        s->analysis_buf[1] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+
+        for (k = 1; k < s->analysis_rdft_len/2; k++) {
+            vars[VAR_F] = k * ((double)inlink->sample_rate /(double)s->analysis_rdft_len);
+            s->analysis_buf[2*k] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+            s->analysis_buf[2*k+1] = 0.0;
+        }
+
+        av_rdft_calc(s->analysis_irdft, s->analysis_buf);
+        center = s->fir_len / 2;
+
+        for (k = 0; k <= center; k++) {
+            double u = k * (M_PI/center);
+            double win;
+            switch (s->wfunc) {
+            case WFUNC_RECTANGULAR:
+                win = 1.0;
+                break;
+            case WFUNC_HANN:
+                win = 0.5 + 0.5 * cos(u);
+                break;
+            case WFUNC_HAMMING:
+                win = 0.53836 + 0.46164 * cos(u);
+                break;
+            case WFUNC_BLACKMAN:
+                win = 0.48 + 0.5 * cos(u) + 0.02 * cos(2*u);
+                break;
+            case WFUNC_NUTTALL3:
+                win = 0.40897 + 0.5 * cos(u) + 0.09103 * cos(2*u);
+                break;
+            case WFUNC_MNUTTALL3:
+                win = 0.4243801 + 0.4973406 * cos(u) + 0.0782793 * cos(2*u);
+                break;
+            case WFUNC_NUTTALL:
+                win = 0.355768 + 0.487396 * cos(u) + 0.144232 * cos(2*u) + 0.012604 * cos(3*u);
+                break;
+            case WFUNC_BNUTTALL:
+                win = 0.3635819 + 0.4891775 * cos(u) + 0.1365995 * cos(2*u) + 0.0106411 * cos(3*u);
+                break;
+            case WFUNC_BHARRIS:
+                win = 0.35875 + 0.48829 * cos(u) + 0.14128 * cos(2*u) + 0.01168 * cos(3*u);
+                break;
+            default:
+                av_assert0(0);
+            }
+            s->analysis_buf[k] *= (2.0/s->analysis_rdft_len) * (2.0/s->rdft_len) * win;
+        }
+
+        for (k = 0; k < center - k; k++) {
+            float tmp = s->analysis_buf[k];
+            s->analysis_buf[k] = s->analysis_buf[center - k];
+            s->analysis_buf[center - k] = tmp;
+        }
+
+        for (k = 1; k <= center; k++)
+            s->analysis_buf[center + k] = s->analysis_buf[center - k];
+
+        memset(s->analysis_buf + s->fir_len, 0, (s->rdft_len - s->fir_len) * sizeof(*s->analysis_buf));
+        av_rdft_calc(s->rdft, s->analysis_buf);
+
+        for (k = 0; k < s->rdft_len; k++) {
+            if (isnan(s->analysis_buf[k]) || isinf(s->analysis_buf[k])) {
+                av_log(ctx, AV_LOG_ERROR, "filter kernel contains nan or infinity.\n");
+                av_expr_free(gain_expr);
+                return AVERROR(EINVAL);
+            }
+        }
+
+        memcpy(s->kernel_tmp_buf + ch * s->rdft_len, s->analysis_buf, s->rdft_len * sizeof(*s->analysis_buf));
+        if (!s->multi)
+            break;
+    }
+
+    memcpy(s->kernel_buf, s->kernel_tmp_buf, (s->multi ? inlink->channels : 1) * s->rdft_len * sizeof(*s->kernel_buf));
+    av_expr_free(gain_expr);
+    return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    FIREqualizerContext *s = ctx->priv;
+    int rdft_bits;
+
+    common_uninit(s);
+
+    s->next_pts = 0;
+    s->frame_nsamples_max = 0;
+
+    s->fir_len = FFMAX(2 * (int)(inlink->sample_rate * s->delay) + 1, 3);
+    s->remaining = s->fir_len - 1;
+
+    for (rdft_bits = RDFT_BITS_MIN; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
+        s->rdft_len = 1 << rdft_bits;
+        s->nsamples_max = s->rdft_len - s->fir_len + 1;
+        if (s->nsamples_max * 2 >= s->fir_len)
+            break;
+    }
+
+    if (rdft_bits > RDFT_BITS_MAX) {
+        av_log(ctx, AV_LOG_ERROR, "too large delay, please decrease it.\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (!(s->rdft = av_rdft_init(rdft_bits, DFT_R2C)) || !(s->irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
+        return AVERROR(ENOMEM);
+
+    for ( ; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
+        s->analysis_rdft_len = 1 << rdft_bits;
+        if (inlink->sample_rate <= s->accuracy * s->analysis_rdft_len)
+            break;
+    }
+
+    if (rdft_bits > RDFT_BITS_MAX) {
+        av_log(ctx, AV_LOG_ERROR, "too small accuracy, please increase it.\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (!(s->analysis_irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
+        return AVERROR(ENOMEM);
+
+    s->analysis_buf = av_malloc_array(s->analysis_rdft_len, sizeof(*s->analysis_buf));
+    s->kernel_tmp_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_tmp_buf));
+    s->kernel_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_buf));
+    s->conv_buf   = av_calloc(2 * s->rdft_len * inlink->channels, sizeof(*s->conv_buf));
+    s->conv_idx   = av_calloc(inlink->channels, sizeof(*s->conv_idx));
+    if (!s->analysis_buf || !s->kernel_tmp_buf || !s->kernel_buf || !s->conv_buf || !s->conv_idx)
+        return AVERROR(ENOMEM);
+
+    av_log(ctx, AV_LOG_DEBUG, "sample_rate = %d, channels = %d, analysis_rdft_len = %d, rdft_len = %d, fir_len = %d, nsamples_max = %d.\n",
+           inlink->sample_rate, inlink->channels, s->analysis_rdft_len, s->rdft_len, s->fir_len, s->nsamples_max);
+
+    if (s->fixed)
+        inlink->min_samples = inlink->max_samples = inlink->partial_buf_size = s->nsamples_max;
+
+    return generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain,
+                           s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+    AVFilterContext *ctx = inlink->dst;
+    FIREqualizerContext *s = ctx->priv;
+    int ch;
+
+    for (ch = 0; ch < inlink->channels; ch++) {
+        fast_convolute(s, s->kernel_buf + (s->multi ? ch * s->rdft_len : 0),
+                       s->conv_buf + 2 * ch * s->rdft_len, s->conv_idx + ch,
+                       (float *) frame->extended_data[ch], frame->nb_samples);
+    }
+
+    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, av_make_q(1, inlink->sample_rate), inlink->time_base);
+    s->frame_nsamples_max = FFMAX(s->frame_nsamples_max, frame->nb_samples);
+    return ff_filter_frame(ctx->outputs[0], frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    FIREqualizerContext *s= ctx->priv;
+    int ret;
+
+    ret = ff_request_frame(ctx->inputs[0]);
+    if (ret == AVERROR_EOF && s->remaining > 0 && s->frame_nsamples_max > 0) {
+        AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(s->remaining, s->frame_nsamples_max));
+
+        if (!frame)
+            return AVERROR(ENOMEM);
+
+        av_samples_set_silence(frame->extended_data, 0, frame->nb_samples, outlink->channels, frame->format);
+        frame->pts = s->next_pts;
+        s->remaining -= frame->nb_samples;
+        ret = filter_frame(ctx->inputs[0], frame);
+    }
+
+    return ret;
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+                           char *res, int res_len, int flags)
+{
+    FIREqualizerContext *s = ctx->priv;
+    int ret = AVERROR(ENOSYS);
+
+    if (!strcmp(cmd, "gain")) {
+        char *gain_cmd;
+
+        gain_cmd = av_strdup(args);
+        if (!gain_cmd)
+            return AVERROR(ENOMEM);
+
+        ret = generate_kernel(ctx, gain_cmd, s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
+        if (ret >= 0) {
+            av_freep(&s->gain_cmd);
+            s->gain_cmd = gain_cmd;
+        } else {
+            av_freep(&gain_cmd);
+        }
+    } else if (!strcmp(cmd, "gain_entry")) {
+        char *gain_entry_cmd;
+
+        gain_entry_cmd = av_strdup(args);
+        if (!gain_entry_cmd)
+            return AVERROR(ENOMEM);
+
+        ret = generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain, gain_entry_cmd);
+        if (ret >= 0) {
+            av_freep(&s->gain_entry_cmd);
+            s->gain_entry_cmd = gain_entry_cmd;
+        } else {
+            av_freep(&gain_entry_cmd);
+        }
+    }
+
+    return ret;
+}
+
+static const AVFilterPad firequalizer_inputs[] = {
+    {
+        .name           = "default",
+        .config_props   = config_input,
+        .filter_frame   = filter_frame,
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .needs_writable = 1,
+    },
+    { NULL }
+};
+
+static const AVFilterPad firequalizer_outputs[] = {
+    {
+        .name           = "default",
+        .request_frame  = request_frame,
+        .type           = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_firequalizer = {
+    .name               = "firequalizer",
+    .description        = NULL_IF_CONFIG_SMALL("Finite Impulse Response Equalizer"),
+    .uninit             = uninit,
+    .query_formats      = query_formats,
+    .process_command    = process_command,
+    .priv_size          = sizeof(FIREqualizerContext),
+    .inputs             = firequalizer_inputs,
+    .outputs            = firequalizer_outputs,
+    .priv_class         = &firequalizer_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ed526493f8..3163831a8b 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -101,6 +101,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(EBUR128,        ebur128,        af);
     REGISTER_FILTER(EQUALIZER,      equalizer,      af);
     REGISTER_FILTER(EXTRASTEREO,    extrastereo,    af);
+    REGISTER_FILTER(FIREQUALIZER,   firequalizer,   af);
     REGISTER_FILTER(FLANGER,        flanger,        af);
     REGISTER_FILTER(HIGHPASS,       highpass,       af);
     REGISTER_FILTER(JOIN,           join,           af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 4a462e7097..480c464cff 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   6
-#define LIBAVFILTER_VERSION_MINOR  34
+#define LIBAVFILTER_VERSION_MINOR  35
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \