[FFmpeg-devel] [PATCH] Port biquads filters from SoX

Stefano Sabatini stefasab at gmail.com
Wed Jan 30 00:31:00 CET 2013


On date Monday 2013-01-28 20:04:19 +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>
> ---
>  libavfilter/Makefile     |   9 +
>  libavfilter/af_biquads.c | 498 +++++++++++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c |   9 +
>  3 files changed, 516 insertions(+)
>  create mode 100644 libavfilter/af_biquads.c

Reminder: minor bump and changelog entry

Also missing docs (you could copy-paste and edit the excellent SoX
documentation).

> 
> 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..26fbc16
> --- /dev/null
> +++ b/libavfilter/af_biquads.c
> @@ -0,0 +1,498 @@
> +/*
> + * 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 "audio.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +enum filter_type {
> +    biquad,
> +    equalizer,
> +    bass,
> +    treble,
> +    band,
> +    bandpass,
> +    bandreject,
> +    allpass,
> +    highpass,
> +    lowpass,
> +};
> +
> +typedef struct chan_cache {
> +    double i1, i2;
> +    double o1, o2;
> +} chan_cache;
> +
> +typedef struct {
> +    const AVClass *class;
> +
> +    int filter_type;
> +    int poles;
> +    int csg;
> +
> +    double gain;
> +    double frequency;
> +    double width;
> +
> +    double a0, a1, a2;
> +    double b0, b1, b2;
> +
> +    chan_cache *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;
> +
> +    return 0;

return av_set_options_string(...);

> +}
> +
> +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;
> +}

Note: we need an helper to reduce the boilerplate.

> +
> +#define BIQUAD_FILTER(name, type)                                             \
> +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;                                                             \
> +        obuf[i] = o0;                                                         \
> +    }                                                                         \
> +}
> +
> +BIQUAD_FILTER(s16, int16_t)
> +BIQUAD_FILTER(s32, int32_t)
> +BIQUAD_FILTER(flt, float)
> +BIQUAD_FILTER(dbl, double)
> +
> +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 = sin(w0) / (2 * p->frequency / p->width);
> +
> +    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);
> +        return AVERROR(EINVAL);
> +    }
> +
> +    switch (p->filter_type) {
> +    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;
> +    }
> +
> +    p->a1 /= p->a0;
> +    p->a2 /= p->a0;
> +    p->b0 /= p->a0;
> +    p->b1 /= p->a0;
> +    p->b2 /= p->a0;
> +
> +    p->cache = av_realloc(p->cache, sizeof(chan_cache) * 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;

assert otherwise

> +    }
> +
> +    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", "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 */

Nit: add an empty line to mark the end of a filter template, same below.

> +#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", "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=202.99}, 0, 99999, FLAGS},
> +    {"w",     "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=202.99}, 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, "Bost or cut lower frequencies.");

Boost?

> +#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", "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=202.99}, 0, 99999, FLAGS},
> +    {"w",     "set shelf transition steep", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=202.99}, 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},
> +};

I'm a bit unhappy with the options duplication (set lut* to see how to
avoid that), OTOH I see the point of having them separated.

> +
> +DEFINE_BIQUAD_FILTER(treble, "Bost or cut upper frequencies.");

Ditto.

> +#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", "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", "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", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 0, 99999, FLAGS},
> +    {"w",     "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 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", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 0, 99999, FLAGS},
> +    {"w",     "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=707.1}, 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", "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},

uhm no help?

[...]

I'd prefer to leave review of the internal code to someone with more
signal processing expertise (Rob?).
-- 
FFmpeg = Fostering & Fast Mortal Perennial Elastic Geek


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