[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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