[FFmpeg-devel] [PATCH] Port biquads filters from SoX
Stefano Sabatini
stefasab at gmail.com
Thu Jan 31 01:54:21 CET 2013
On date Wednesday 2013-01-30 20:17:40 +0000, Paul B Mahol encoded:
> Adds allpass, bandpass, bandreject, bass, biquad,
> equalizer, highpass, lowpass and treble filter.
>
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> doc/filters.texi | 298 ++++++++++++++++++++++++
> libavfilter/Makefile | 9 +
> libavfilter/af_biquads.c | 589 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 9 +
> 4 files changed, 905 insertions(+)
> create mode 100644 libavfilter/af_biquads.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 21e2cff..f553992 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -282,6 +282,304 @@ aconvert=u8:auto
> @end example
> @end itemize
>
> + at section allpass
> +
> +Apply a two-pole all-pass filter with central frequency (in Hz)
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +A frequency in Hz.
Nit: Set frequency in Hz.
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Here it seems you inverted name and description.
> +
> + at item width, w
> +Used to specify the band-width of a filter in width_type units.
Specify the band-width... for grammar consistency.
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +A frequency in Hz. Default is 3000.
Set a frequency ...
> +
> + at item poles, p
> +Number of poles. Default is 2.
Set number of poles ...
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Ditto
> +
> + at item width, w
> +Applies only to double-pole filter.
Set width ... Apply only ...
> +The default is 0.707 Q and gives a Butterworth response.
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +A frequency in Hz. Default is 500.
> +
> + at item poles, p
> +Number of poles. Default is 2.
Ditto
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Same
> +
> + at item width, w
> +Applies only to double-pole filter.
Set width ... Apply only ...
> +The default is 0.707 Q and gives a Butterworth response.
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item gain, g
> +Gives the gain at 0 Hz. Its useful range is about -20
Give
> +(for a large cut) to +20 (for a large boost).
> +Beware of Clipping when using a positive gain.
_c_lipping?
> +
> + at item frequency, f
> +Sets the filter's central frequency and so can be used
Set
> +to extend or reduce the frequency range to be boosted or cut.
> +The default value is @code{100} Hz.
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Ditto.
> +
> + at item width, w
> +Determines how steep is the filter's shelf transition.
Determine...
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item gain, g
> +Gives the gain at whichever is the lower of ~22 kHz and the
Give
> +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.
> +
> + at item frequency, f
> +Sets the filter's central frequency and so can be used
Set
> +to extend or reduce the frequency range to be boosted or cut.
> +The default value is @code{3000} Hz.
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Ditto
> +
> + at item width, w
> +Determines how steep is the filter's shelf transition.
Determine
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +Sets the filter's central frequency. Default is @code{3000}.
Set
> +
> + at item csg
> +Use constant skirt gain. Defaults to disabled.
... skirt gain if set to 1. Default to 0.
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
> +
> + at item width, w
> +Used to specify the band-width of a filter in Hz.
Specify the ...
> + at end table
> +
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +Sets the filter's central frequency. Default is @code{3000}.
Set
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Ditto
> +
> + at item width, w
> +Used to specify the band-width of a filter in Hz.
Specify
> + at end table
> +
> + at 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.
> +
> + at 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.
> +
> + at table @option
> + at item frequency, f
> +Gives the filter's central frequency in Hz.
Give
> +
> + at item width_type
> +Set method to specify band-width of filter.
> + at table @option
> + at item @var{Hz}
> + at code{h}
> + at item @var{Q-Factor}
> + at code{q}
> + at item @var{octave}
> + at code{o}
> + at item @var{slope}
> + at code{s}
> + at end table
Ditto
> +
> + at item width, w
> +The band-width.
Set the band-width?
> +
> + at item gain, g
> +The required gain or attenuation in dB.
Set the required ...
> +Beware of clipping when using a positive gain.
> + at end table
> +
> @section afade
>
> Apply fade-in/out effect to input audio.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 5835a7e..938b183 100644
> --- a/libavfilter/Makefile
> +++ b/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
>
> diff --git a/libavfilter/af_biquads.c b/libavfilter/af_biquads.c
> new file mode 100644
> index 0000000..fd4244b
> --- /dev/null
> +++ b/libavfilter/af_biquads.c
> @@ -0,0 +1,589 @@
> +/*
> + * Copyright (c) 2013 Paul B Mahol
> + * Copyright (c) 2006-2008 Rob Sykes <robs at 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 at audioimagination.com>
> + * see http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
> + *
> + * 1-pole filters based on code (c) 2000 Chris Bagwell <cbagwell at 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, "frequency and/or width <= 0\n");
> + 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 *i1, double *i2, double *o1, double *o2, \
> + double b0, double b1, double b2, \
> + double a1, double a2) \
> +{ \
> + const type *ibuf = input; \
> + type *obuf = output; \
> + 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; \
> + } \
> + } \
> +}
> +
> +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,
> + "frequency %f must be less than half the sample-rate %d\n",
> + p->frequency, inlink->sample_rate);
Nit: Invalid frequency %f. Frequency must be ...
[...]
No more nits from me. Feel free to push if you tested it enough and
there are no people willing to review the processing code.
Thanks.
--
FFmpeg = Faithless Fast Mournful Patchable Encoding/decoding Gladiator
More information about the ffmpeg-devel
mailing list