Port biquads filters from SoX

Adds allpass, bass, bandpass, bandreject, biquad,
equalizer, highpass, lowpass and treble filter.

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Changelog

@@ -10,6 +10,8 @@ version <next>:
 - EVRC decoder
 - audio fade filter
 - filtering audio with unknown channel layout
+- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
+  and treble audio filter
 
 
 version 1.1:

doc/filters.texi

@@ -282,6 +282,274 @@ aconvert=u8:auto
 @end example
 @end itemize
 
+@section allpass
+
+Apply a two-pole all-pass filter with central frequency (in Hz)
+@var{frequency}, and filter-width @var{width}.
+An all-pass filter changes the audio's frequency to phase relationship
+without changing its frequency to amplitude relationship.
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set frequency in Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+@end table
+
+@section highpass
+
+Apply a high-pass filter with 3dB point frequency.
+The filter can be either single-pole, or double-pole (the default).
+The filter roll off at 6dB per pole per octave (20dB per pole per decade).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set frequency in Hz. Default is 3000.
+
+@item poles, p
+Set number of poles. Default is 2.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+Applies only to double-pole filter.
+The default is 0.707q and gives a Butterworth response.
+@end table
+
+@section lowpass
+
+Apply a low-pass filter with 3dB point frequency.
+The filter can be either single-pole or double-pole (the default).
+The filter roll off at 6dB per pole per octave (20dB per pole per decade).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set frequency in Hz. Default is 500.
+
+@item poles, p
+Set number of poles. Default is 2.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+Applies only to double-pole filter.
+The default is 0.707q and gives a Butterworth response.
+@end table
+
+@section bass
+
+Boost or cut the bass (lower) frequencies of the audio using a two-pole
+shelving filter with a response similar to that of a standard
+hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item gain, g
+Give the gain at 0 Hz. Its useful range is about -20
+(for a large cut) to +20 (for a large boost).
+Beware of clipping when using a positive gain.
+
+@item frequency, f
+Set the filter's central frequency and so can be used
+to extend or reduce the frequency range to be boosted or cut.
+The default value is @code{100} Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Determine how steep is the filter's shelf transition.
+@end table
+
+@section treble
+
+Boost or cut treble (upper) frequencies of the audio using a two-pole
+shelving filter with a response similar to that of a standard
+hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item gain, g
+Give the gain at whichever is the lower of ~22 kHz and the
+Nyquist frequency. Its useful range is about -20 (for a large cut)
+to +20 (for a large boost). Beware of clipping when using a positive gain.
+
+@item frequency, f
+Set the filter's central frequency and so can be used
+to extend or reduce the frequency range to be boosted or cut.
+The default value is @code{3000} Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Determine how steep is the filter's shelf transition.
+@end table
+
+@section bandpass
+
+Apply a two-pole Butterworth band-pass filter with central
+frequency @var{frequency}, and (3dB-point) band-width width.
+The @var{csg} option selects a constant skirt gain (peak gain = Q)
+instead of the default: constant 0dB peak gain.
+The filter roll off at 6dB per octave (20dB per decade).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set the filter's central frequency. Default is @code{3000}.
+
+@item csg
+Constant skirt gain if set to 1. Defaults to 0.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+@end table
+
+@section bandreject
+
+Apply a two-pole Butterworth band-reject filter with central
+frequency @var{frequency}, and (3dB-point) band-width @var{width}.
+The filter roll off at 6dB per octave (20dB per decade).
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set the filter's central frequency. Default is @code{3000}.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+@end table
+
+@section biquad
+
+Apply a biquad IIR filter with the given coefficients.
+Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
+are the numerator and denominator coefficients respectively.
+
+@section equalizer
+
+Apply a two-pole peaking equalisation (EQ) filter. With this
+filter, the signal-level at and around a selected frequency can
+be increased or decreased, whilst (unlike bandpass and bandreject
+filters) that at all other frequencies is unchanged.
+
+In order to produce complex equalisation curves, this filter can
+be given several times, each with a different central frequency.
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+@table @option
+@item frequency, f
+Set the filter's central frequency in Hz.
+
+@item width_type
+Set method to specify band-width of filter.
+@table @option
+@item @var{h} (Hz)
+@item @var{q} (Q-Factor)
+@item @var{o} (octave)
+@item @var{s} (slope)
+@end table
+
+@item width, w
+Specify the band-width of a filter in width_type units.
+
+@item gain, g
+Set the required gain or attenuation in dB.
+Beware of clipping when using a positive gain.
+@end table
+
 @section afade
 
 Apply fade-in/out effect to input audio.

libavfilter/Makefile

@@ -53,6 +53,7 @@ OBJS-$(CONFIG_SWSCALE)                       += lswsutils.o
 OBJS-$(CONFIG_ACONVERT_FILTER)               += af_aconvert.o
 OBJS-$(CONFIG_AFADE_FILTER)                  += af_afade.o
 OBJS-$(CONFIG_AFORMAT_FILTER)                += af_aformat.o
+OBJS-$(CONFIG_ALLPASS_FILTER)                += af_biquads.o
 OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
@@ -68,14 +69,22 @@ OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(CONFIG_ASTREAMSYNC_FILTER)            += af_astreamsync.o
 OBJS-$(CONFIG_ASYNCTS_FILTER)                += af_asyncts.o
 OBJS-$(CONFIG_ATEMPO_FILTER)                 += af_atempo.o
+OBJS-$(CONFIG_BANDPASS_FILTER)               += af_biquads.o
+OBJS-$(CONFIG_BANDREJECT_FILTER)             += af_biquads.o
+OBJS-$(CONFIG_BASS_FILTER)                   += af_biquads.o
+OBJS-$(CONFIG_BIQUAD_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_CHANNELMAP_FILTER)             += af_channelmap.o
 OBJS-$(CONFIG_CHANNELSPLIT_FILTER)           += af_channelsplit.o
 OBJS-$(CONFIG_EARWAX_FILTER)                 += af_earwax.o
 OBJS-$(CONFIG_EBUR128_FILTER)                += f_ebur128.o
+OBJS-$(CONFIG_EQUALIZER_FILTER)              += af_biquads.o
+OBJS-$(CONFIG_HIGHPASS_FILTER)               += af_biquads.o
 OBJS-$(CONFIG_JOIN_FILTER)                   += af_join.o
+OBJS-$(CONFIG_LOWPASS_FILTER)                += af_biquads.o
 OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o
 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o
 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o
+OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o
 OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume.o
 OBJS-$(CONFIG_VOLUMEDETECT_FILTER)           += af_volumedetect.o
 

libavfilter/af_biquads.c

@@ -0,0 +1,599 @@
+/*
+ * Copyright (c) 2013 Paul B Mahol
+ * Copyright (c) 2006-2008 Rob Sykes <robs@users.sourceforge.net>
+ *
+ * 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
+ */
+
+/*
+ * 2-pole filters designed by Robert Bristow-Johnson <rbj@audioimagination.com>
+ *   see http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
+ *
+ * 1-pole filters based on code (c) 2000 Chris Bagwell <cbagwell@sprynet.com>
+ *   Algorithms: Recursive single pole low/high pass filter
+ *   Reference: The Scientist and Engineer's Guide to Digital Signal Processing
+ *
+ *   low-pass: output[N] = input[N] * A + output[N-1] * B
+ *     X = exp(-2.0 * pi * Fc)
+ *     A = 1 - X
+ *     B = X
+ *     Fc = cutoff freq / sample rate
+ *
+ *     Mimics an RC low-pass filter:
+ *
+ *     ---/\/\/\/\----------->
+ *                   |
+ *                  --- C
+ *                  ---
+ *                   |
+ *                   |
+ *                   V
+ *
+ *   high-pass: output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
+ *     X  = exp(-2.0 * pi * Fc)
+ *     A0 = (1 + X) / 2
+ *     A1 = -(1 + X) / 2
+ *     B1 = X
+ *     Fc = cutoff freq / sample rate
+ *
+ *     Mimics an RC high-pass filter:
+ *
+ *         || C
+ *     ----||--------->
+ *         ||    |
+ *               <
+ *               > R
+ *               <
+ *               |
+ *               V
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/avassert.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+enum FilterType {
+    biquad,
+    equalizer,
+    bass,
+    treble,
+    band,
+    bandpass,
+    bandreject,
+    allpass,
+    highpass,
+    lowpass,
+};
+
+enum WidthType {
+    NONE,
+    HZ,
+    OCTAVE,
+    QFACTOR,
+    SLOPE,
+};
+
+typedef struct ChanCache {
+    double i1, i2;
+    double o1, o2;
+} ChanCache;
+
+typedef struct {
+    const AVClass *class;
+
+    enum FilterType filter_type;
+    enum WidthType width_type;
+    int poles;
+    int csg;
+
+    double gain;
+    double frequency;
+    double width;
+
+    double a0, a1, a2;
+    double b0, b1, b2;
+
+    ChanCache *cache;
+
+    void (*filter)(const void *ibuf, void *obuf, int len,
+                   double *i1, double *i2, double *o1, double *o2,
+                   double b0, double b1, double b2, double a1, double a2);
+} BiquadsContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args)
+{
+    BiquadsContext *p = ctx->priv;
+    int ret;
+
+    av_opt_set_defaults(p);
+
+    if ((ret = av_set_options_string(p, args, "=", ":")) < 0)
+        return ret;
+
+    if (p->filter_type != biquad) {
+        if (p->frequency <= 0 || p->width <= 0) {
+            av_log(ctx, AV_LOG_ERROR, "Invalid frequency %f and/or width %f <= 0\n",
+                   p->frequency, p->width);
+            return AVERROR(EINVAL);
+        }
+    }
+
+    return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AVFilterFormats *formats;
+    AVFilterChannelLayouts *layouts;
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_S16P,
+        AV_SAMPLE_FMT_S32P,
+        AV_SAMPLE_FMT_FLTP,
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+
+    layouts = ff_all_channel_layouts();
+    if (!layouts)
+        return AVERROR(ENOMEM);
+    ff_set_common_channel_layouts(ctx, layouts);
+
+    formats = ff_make_format_list(sample_fmts);
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_formats(ctx, formats);
+
+    formats = ff_all_samplerates();
+    if (!formats)
+        return AVERROR(ENOMEM);
+    ff_set_common_samplerates(ctx, formats);
+
+    return 0;
+}
+
+#define BIQUAD_FILTER(name, type, min, max)                                   \
+static void biquad_## name (const void *input, void *output, int len,         \
+                            double *in1, double *in2,                         \
+                            double *out1, double *out2,                       \
+                            double b0, double b1, double b2,                  \
+                            double a1, double a2)                             \
+{                                                                             \
+    const type *ibuf = input;                                                 \
+    type *obuf = output;                                                      \
+    double i1 = *in1;                                                         \
+    double i2 = *in2;                                                         \
+    double o1 = *out1;                                                        \
+    double o2 = *out2;                                                        \
+    int i;                                                                    \
+                                                                              \
+    for (i = 0; i < len; i++) {                                               \
+        double o0 = ibuf[i] * b0 + i1 * b1 + i2 * b2 - o1 * a1 - o2 * a2;     \
+        i2 = i1;                                                              \
+        i1 = ibuf[i];                                                         \
+        o2 = o1;                                                              \
+        o1 = o0;                                                              \
+        if (o0 < min) {                                                       \
+            av_log(NULL, AV_LOG_WARNING, "clipping\n");                       \
+            obuf[i] = min;                                                    \
+        } else if (o0 > max) {                                                \
+            av_log(NULL, AV_LOG_WARNING, "clipping\n");                       \
+            obuf[i] = max;                                                    \
+        } else {                                                              \
+            obuf[i] = o0;                                                     \
+        }                                                                     \
+    }                                                                         \
+    *in1  = i1;                                                               \
+    *in2  = i2;                                                               \
+    *out1 = o1;                                                               \
+    *out2 = o2;                                                               \
+}
+
+BIQUAD_FILTER(s16, int16_t, INT16_MIN, INT16_MAX)
+BIQUAD_FILTER(s32, int32_t, INT32_MIN, INT32_MAX)
+BIQUAD_FILTER(flt, float,   -1., 1.)
+BIQUAD_FILTER(dbl, double,  -1., 1.)
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx    = outlink->src;
+    BiquadsContext *p       = ctx->priv;
+    AVFilterLink *inlink    = ctx->inputs[0];
+    double A = exp(p->gain / 40 * log(10.));
+    double w0 = 2 * M_PI * p->frequency / inlink->sample_rate;
+    double alpha;
+
+    if (w0 > M_PI) {
+        av_log(ctx, AV_LOG_ERROR,
+               "Invalid frequency %f. Frequency must be less than half the sample-rate %d.\n",
+               p->frequency, inlink->sample_rate);
+        return AVERROR(EINVAL);
+    }
+
+    switch (p->width_type) {
+    case NONE:
+        alpha = 0.0;
+        break;
+    case HZ:
+        alpha = sin(w0) / (2 * p->frequency / p->width);
+        break;
+    case OCTAVE:
+        alpha = sin(w0) * sinh(log(2.) / 2 * p->width * w0 / sin(w0));
+        break;
+    case QFACTOR:
+        alpha = sin(w0) / (2 * p->width);
+        break;
+    case SLOPE:
+        alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / p->width - 1) + 2);
+        break;
+    default:
+        av_assert0(0);
+    }
+
+    switch (p->filter_type) {
+    case biquad:
+        break;
+    case equalizer:
+        p->a0 =   1 + alpha / A;
+        p->a1 =  -2 * cos(w0);
+        p->a2 =   1 - alpha / A;
+        p->b0 =   1 + alpha * A;
+        p->b1 =  -2 * cos(w0);
+        p->b2 =   1 - alpha * A;
+        break;
+    case bass:
+        p->a0 =          (A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha;
+        p->a1 =    -2 * ((A - 1) + (A + 1) * cos(w0));
+        p->a2 =          (A + 1) + (A - 1) * cos(w0) - 2 * sqrt(A) * alpha;
+        p->b0 =     A * ((A + 1) - (A - 1) * cos(w0) + 2 * sqrt(A) * alpha);
+        p->b1 = 2 * A * ((A - 1) - (A + 1) * cos(w0));
+        p->b2 =     A * ((A + 1) - (A - 1) * cos(w0) - 2 * sqrt(A) * alpha);
+        break;
+    case treble:
+        p->a0 =          (A + 1) - (A - 1) * cos(w0) + 2 * sqrt(A) * alpha;
+        p->a1 =     2 * ((A - 1) - (A + 1) * cos(w0));
+        p->a2 =          (A + 1) - (A - 1) * cos(w0) - 2 * sqrt(A) * alpha;
+        p->b0 =     A * ((A + 1) + (A - 1) * cos(w0) + 2 * sqrt(A) * alpha);
+        p->b1 =-2 * A * ((A - 1) + (A + 1) * cos(w0));
+        p->b2 =     A * ((A + 1) + (A - 1) * cos(w0) - 2 * sqrt(A) * alpha);
+        break;
+    case bandpass:
+        if (p->csg) {
+            p->a0 =  1 + alpha;
+            p->a1 = -2 * cos(w0);
+            p->a2 =  1 - alpha;
+            p->b0 =  sin(w0) / 2;
+            p->b1 =  0;
+            p->b2 = -sin(w0) / 2;
+        } else {
+            p->a0 =  1 + alpha;
+            p->a1 = -2 * cos(w0);
+            p->a2 =  1 - alpha;
+            p->b0 =  alpha;
+            p->b1 =  0;
+            p->b2 = -alpha;
+        }
+        break;
+    case bandreject:
+        p->a0 =  1 + alpha;
+        p->a1 = -2 * cos(w0);
+        p->a2 =  1 - alpha;
+        p->b0 =  1;
+        p->b1 = -2 * cos(w0);
+        p->b2 =  1;
+        break;
+    case lowpass:
+        if (p->poles == 1) {
+            p->a0 = 1;
+            p->a1 = -exp(-w0);
+            p->a2 = 0;
+            p->b0 = 1 + p->a1;
+            p->b1 = 0;
+            p->b2 = 0;
+        } else {
+            p->a0 =  1 + alpha;
+            p->a1 = -2 * cos(w0);
+            p->a2 =  1 - alpha;
+            p->b0 = (1 - cos(w0)) / 2;
+            p->b1 =  1 - cos(w0);
+            p->b2 = (1 - cos(w0)) / 2;
+        }
+        break;
+    case highpass:
+        if (p->poles == 1) {
+            p->a0 = 1;
+            p->a1 = -exp(-w0);
+            p->a2 = 0;
+            p->b0 = (1 - p->a1) / 2;
+            p->b1 = -p->b0;
+            p->b2 = 0;
+        } else {
+            p->a0 =   1 + alpha;
+            p->a1 =  -2 * cos(w0);
+            p->a2 =   1 - alpha;
+            p->b0 =  (1 + cos(w0)) / 2;
+            p->b1 = -(1 + cos(w0));
+            p->b2 =  (1 + cos(w0)) / 2;
+        }
+        break;
+    case allpass:
+        p->a0 =  1 + alpha;
+        p->a1 = -2 * cos(w0);
+        p->a2 =  1 - alpha;
+        p->b0 =  1 - alpha;
+        p->b1 = -2 * cos(w0);
+        p->b2 =  1 + alpha;
+        break;
+    default:
+        av_assert0(0);
+    }
+
+    p->a1 /= p->a0;
+    p->a2 /= p->a0;
+    p->b0 /= p->a0;
+    p->b1 /= p->a0;
+    p->b2 /= p->a0;
+
+    p->cache = av_realloc_f(p->cache, sizeof(ChanCache), inlink->channels);
+    if (!p->cache)
+        return AVERROR(ENOMEM);
+
+    switch (inlink->format) {
+    case AV_SAMPLE_FMT_S16P: p->filter = biquad_s16; break;
+    case AV_SAMPLE_FMT_S32P: p->filter = biquad_s32; break;
+    case AV_SAMPLE_FMT_FLTP: p->filter = biquad_flt; break;
+    case AV_SAMPLE_FMT_DBLP: p->filter = biquad_dbl; break;
+    default: av_assert0(0);
+    }
+
+    return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    BiquadsContext *p       = inlink->dst->priv;
+    AVFilterLink *outlink   = inlink->dst->outputs[0];
+    AVFilterBufferRef *out_buf;
+    int nb_samples = buf->audio->nb_samples;
+    int ch;
+
+    if (buf->perms & AV_PERM_WRITE) {
+        out_buf = buf;
+    } else {
+        out_buf = ff_get_audio_buffer(inlink, AV_PERM_WRITE, nb_samples);
+        if (!out_buf)
+            return AVERROR(ENOMEM);
+        out_buf->pts = buf->pts;
+    }
+
+    for (ch = 0; ch < buf->audio->channels; ch++)
+        p->filter((const float *)buf->extended_data[ch],
+                   (float *)out_buf->extended_data[ch], nb_samples,
+                   &p->cache[ch].i1, &p->cache[ch].i2,
+                   &p->cache[ch].o1, &p->cache[ch].o2,
+                   p->b0, p->b1, p->b2, p->a1, p->a2);
+
+    if (buf != out_buf)
+        avfilter_unref_buffer(buf);
+
+    return ff_filter_frame(outlink, out_buf);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    BiquadsContext *p = ctx->priv;
+
+    av_freep(&p->cache);
+    av_opt_free(p);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .filter_frame = filter_frame,
+    },
+    { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_output,
+    },
+    { NULL }
+};
+
+#define OFFSET(x) offsetof(BiquadsContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+#define DEFINE_BIQUAD_FILTER(name_, description_)                       \
+AVFILTER_DEFINE_CLASS(name_);                                           \
+static av_cold int name_##_init(AVFilterContext *ctx, const char *args) \
+{                                                                       \
+    BiquadsContext *p = ctx->priv;                                      \
+    p->class = &name_##_class;                                          \
+    p->filter_type = name_;                                             \
+    return init(ctx, args);                                             \
+}                                                                       \
+                                                         \
+AVFilter avfilter_af_##name_ = {                         \
+    .name          = #name_,                             \
+    .description   = NULL_IF_CONFIG_SMALL(description_), \
+    .priv_size     = sizeof(BiquadsContext),             \
+    .init          = name_##_init,                       \
+    .uninit        = uninit,                             \
+    .query_formats = query_formats,                      \
+    .inputs        = inputs,                             \
+    .outputs       = outputs,                            \
+    .priv_class    = &name_##_class,                     \
+}
+
+#if CONFIG_EQUALIZER_FILTER
+static const AVOption equalizer_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 999, FLAGS},
+    {"w",     "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 999, FLAGS},
+    {"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {"g",    "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(equalizer, "Apply two-pole peaking equalization (EQ) filter.");
+#endif  /* CONFIG_EQUALIZER_FILTER */
+#if CONFIG_BASS_FILTER
+static const AVOption bass_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 99999, FLAGS},
+    {"w",     "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 99999, FLAGS},
+    {"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {"g",    "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(bass, "Boost or cut lower frequencies.");
+#endif  /* CONFIG_BASS_FILTER */
+#if CONFIG_TREBLE_FILTER
+static const AVOption treble_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 99999, FLAGS},
+    {"w",     "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 99999, FLAGS},
+    {"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {"g",    "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(treble, "Boost or cut upper frequencies.");
+#endif  /* CONFIG_TREBLE_FILTER */
+#if CONFIG_BANDPASS_FILTER
+static const AVOption bandpass_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 999, FLAGS},
+    {"w",     "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 999, FLAGS},
+    {"csg",   "use constant skirt gain", OFFSET(csg), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(bandpass, "Apply a two-pole Butterworth band-pass filter.");
+#endif  /* CONFIG_BANDPASS_FILTER */
+#if CONFIG_BANDREJECT_FILTER
+static const AVOption bandreject_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 999, FLAGS},
+    {"w",     "set band-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 999, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(bandreject, "Apply a two-pole Butterworth band-reject filter.");
+#endif  /* CONFIG_BANDREJECT_FILTER */
+#if CONFIG_LOWPASS_FILTER
+static const AVOption lowpass_options[] = {
+    {"frequency", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=500}, 0, 999999, FLAGS},
+    {"f",         "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=500}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.707}, 0, 99999, FLAGS},
+    {"w",     "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.707}, 0, 99999, FLAGS},
+    {"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
+    {"p",     "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(lowpass, "Apply a low-pass filter with 3dB point frequency.");
+#endif  /* CONFIG_LOWPASS_FILTER */
+#if CONFIG_HIGHPASS_FILTER
+static const AVOption highpass_options[] = {
+    {"frequency", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"f",         "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=QFACTOR}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.707}, 0, 99999, FLAGS},
+    {"w",     "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=0.707}, 0, 99999, FLAGS},
+    {"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
+    {"p",     "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(highpass, "Apply a high-pass filter with 3dB point frequency.");
+#endif  /* CONFIG_HIGHPASS_FILTER */
+#if CONFIG_ALLPASS_FILTER
+static const AVOption allpass_options[] = {
+    {"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"f",         "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
+    {"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=HZ}, HZ, SLOPE, FLAGS, "width_type"},
+    {"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HZ}, 0, 0, FLAGS, "width_type"},
+    {"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"},
+    {"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"},
+    {"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"},
+    {"width", "set filter-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 0, 99999, FLAGS},
+    {"w",     "set filter-width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 0, 99999, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(allpass, "Apply a two-pole all-pass filter.");
+#endif  /* CONFIG_ALLPASS_FILTER */
+#if CONFIG_BIQUAD_FILTER
+static const AVOption biquad_options[] = {
+    {"a0", NULL, OFFSET(a0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {"a1", NULL, OFFSET(a1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {"a2", NULL, OFFSET(a2), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {"b0", NULL, OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {"b1", NULL, OFFSET(b1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {"b2", NULL, OFFSET(b2), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT16_MAX, INT16_MAX, FLAGS},
+    {NULL},
+};
+
+DEFINE_BIQUAD_FILTER(biquad, "Apply a biquad IIR filter with the given coefficients.");
+#endif  /* CONFIG_BIQUAD_FILTER */

libavfilter/allfilters.c

@@ -47,6 +47,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ACONVERT,       aconvert,       af);
     REGISTER_FILTER(AFADE,          afade,          af);
     REGISTER_FILTER(AFORMAT,        aformat,        af);
+    REGISTER_FILTER(ALLPASS,        allpass,        af);
     REGISTER_FILTER(AMERGE,         amerge,         af);
     REGISTER_FILTER(AMIX,           amix,           af);
     REGISTER_FILTER(ANULL,          anull,          af);
@@ -62,14 +63,22 @@ void avfilter_register_all(void)
     REGISTER_FILTER(ASTREAMSYNC,    astreamsync,    af);
     REGISTER_FILTER(ASYNCTS,        asyncts,        af);
     REGISTER_FILTER(ATEMPO,         atempo,         af);
+    REGISTER_FILTER(BANDPASS,       bandpass,       af);
+    REGISTER_FILTER(BANDREJECT,     bandreject,     af);
+    REGISTER_FILTER(BASS,           bass,           af);
+    REGISTER_FILTER(BIQUAD,         biquad,         af);
     REGISTER_FILTER(CHANNELMAP,     channelmap,     af);
     REGISTER_FILTER(CHANNELSPLIT,   channelsplit,   af);
     REGISTER_FILTER(EARWAX,         earwax,         af);
     REGISTER_FILTER(EBUR128,        ebur128,        af);
+    REGISTER_FILTER(EQUALIZER,      equalizer,      af);
+    REGISTER_FILTER(HIGHPASS,       highpass,       af);
     REGISTER_FILTER(JOIN,           join,           af);
+    REGISTER_FILTER(LOWPASS,        lowpass,        af);
     REGISTER_FILTER(PAN,            pan,            af);
     REGISTER_FILTER(RESAMPLE,       resample,       af);
     REGISTER_FILTER(SILENCEDETECT,  silencedetect,  af);
+    REGISTER_FILTER(TREBLE,         treble,         af);
     REGISTER_FILTER(VOLUME,         volume,         af);
     REGISTER_FILTER(VOLUMEDETECT,   volumedetect,   af);
 

libavfilter/version.h

@@ -29,8 +29,8 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  3
-#define LIBAVFILTER_VERSION_MINOR  34
+#define LIBAVFILTER_VERSION_MINOR  35
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
                                                LIBAVFILTER_VERSION_MINOR, \