@@ -6,6 +6,7 @@ version <next>:

 - fieldhint filter

 - loop video filter and aloop audio filter

 - Bob Weaver deinterlacing filter

+- firequalizer filter

 version 3.0:

@@ -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

@@ -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"

@@ -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.

@@ -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

new file mode 100644

@@ -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,

+};

@@ -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);

@@ -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, \