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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>
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1 parent ff23b76 commit b70ea49ca32cbff8be5c6f85a94600bb5fcc8ac8 @richardpl richardpl committed Jan 24, 2013
Showing with 889 additions and 2 deletions.
  1. +2 −0 Changelog
  2. +268 −0 doc/filters.texi
  3. +9 −0 libavfilter/Makefile
  4. +599 −0 libavfilter/af_biquads.c
  5. +9 −0 libavfilter/allfilters.c
  6. +2 −2 libavfilter/version.h
View
2 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:
View
268 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.
View
9 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
View
599 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 */
View
9 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);
View
4 libavfilter/version.h
@@ -29,8 +29,8 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
-#define LIBAVFILTER_VERSION_MINOR 34
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR 35
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \

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