[FFmpeg-devel] [PATCH] avfilter: add superequalizer filter
James Almer
jamrial at gmail.com
Sat Jun 17 20:19:10 EEST 2017
On 6/17/2017 1:00 PM, Paul B Mahol wrote:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> libavfilter/Makefile | 1 +
> libavfilter/af_superequalizer.c | 381 ++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 3 files changed, 383 insertions(+)
> create mode 100644 libavfilter/af_superequalizer.c
>
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index 04ec9b8..52c44d2 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -109,6 +109,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
> OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
> OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
> OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
> +OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o
> OBJS-$(CONFIG_SURROUND_FILTER) += af_surround.o
> OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
> OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o
> diff --git a/libavfilter/af_superequalizer.c b/libavfilter/af_superequalizer.c
> new file mode 100644
> index 0000000..c569394
> --- /dev/null
> +++ b/libavfilter/af_superequalizer.c
> @@ -0,0 +1,381 @@
> +/*
> + * 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
> + */
> +
> +#include "libavutil/opt.h"
> +
> +#include "libavcodec/avfft.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "internal.h"
> +
> +#define NBANDS 17
> +#define M 15
> +
> +typedef struct EqParameter {
> + float lower, upper, gain;
> +} EqParameter;
> +
> +typedef struct SuperEqualizerContext {
> + const AVClass *class;
> +
> + EqParameter params[NBANDS+1];
> +
> + float gains[NBANDS+1];
> +
> + float fact[M+1];
> + float aa;
> + float iza;
> + float *lires, *rires, *irest;
> + float *fsamples;
> + int chg_ires, cur_ires;
> + int winlen, winlenbit, tabsize;
> +
> + AVFrame *in, *out;
> + RDFTContext *rdft, *irdft;
> +} SuperEqualizerContext;
> +
> +static const float bands[] = {
> + 65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
> + 1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
> +};
> +
> +static float izero(SuperEqualizerContext *s, float x)
> +{
> + float ret = 1;
> + int m;
> +
> + for (m = 1; m <= M; m++) {
> + float t;
> +
> + t = pow(x / 2, m) / s->fact[m];
powf
> + ret += t*t;
> + }
> +
> + return ret;
> +}
> +
> +static float hn_lpf(int n, float f, float fs)
> +{
> + float t = 1 / fs;
> + float omega = 2 * M_PI * f;
> +
> + if (n * omega * t == 0)
> + return 2 * f * t;
> + return 2 * f * t * sin(n * omega * t)/(n * omega * t);
sinf
> +}
> +
> +static float hn_imp(int n)
> +{
> + return n == 0 ? 1.0 : 0.0;
Use float constants.
Same for every other constant below.
> +}
> +
> +static float hn(int n, EqParameter *param, float fs)
> +{
> + float ret, lhn;
> + int i;
> +
> + lhn = hn_lpf(n, param[0].upper, fs);
> + ret = param[0].gain*lhn;
> +
> + for (i = 1; i < NBANDS+1 && param[i].upper < fs / 2; i++) {
> + float lhn2 = hn_lpf(n, param[i].upper, fs);
> + ret += param[i].gain * (lhn2-lhn);
> + lhn = lhn2;
> + }
> +
> + ret += param[i].gain * (hn_imp(n) - lhn);
> +
> + return ret;
> +}
> +
> +static float alpha(float a)
> +{
> + if (a <= 21)
> + return 0;
> + if (a <= 50)
> + return .5842 * pow(a - 21,0.4) + 0.07886 * (a - 21);
> + return .1102 * (a - 8.7);
> +}
> +
> +static float win(SuperEqualizerContext *s, float n, int N)
> +{
> + return izero(s, alpha(s->aa) * sqrt(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
sqrtf
All these are called inside loops so it's best to prevent the compiler
from pointlessly using doubles.
> +}
> +
> +static void
> +process_param(float *bc, EqParameter *param, float fs, int ch)
> +{
> + int i;
> +
> + for (i = 0; i <= NBANDS; i++) {
> + param[i].lower = i == 0 ? 0 : bands[i - 1];
> + param[i].upper = i == NBANDS - 1 ? fs : bands[i];
> + param[i].gain = bc[i];
> + }
> +}
> +
> +static int equ_init(SuperEqualizerContext *s, int wb)
> +{
> + int i,j;
> +
> + s->rdft = av_rdft_init(wb, DFT_R2C);
> + s->irdft = av_rdft_init(wb, IDFT_C2R);
> + if (!s->rdft || !s->irdft)
> + return AVERROR(ENOMEM);
> +
> + s->aa = 96;
> + s->winlen = (1 << (wb-1))-1;
> + s->winlenbit = wb;
> + s->tabsize = 1 << wb;
> +
> + s->lires = av_calloc(s->tabsize, sizeof(float));
> + s->rires = av_calloc(s->tabsize, sizeof(float));
> + s->irest = av_calloc(s->tabsize, sizeof(float));
> + s->fsamples = av_calloc(s->tabsize, sizeof(float));
Missing check for allocation failure. Also, none of these seem to be
freed during uninit.
> +
> + s->cur_ires = 1;
> + s->chg_ires = 1;
> +
> + for (i = 0; i <= M; i++) {
> + s->fact[i] = 1;
> + for (j = 1; j <= i; j++)
> + s->fact[i] *= j;
> + }
> +
> + s->iza = izero(s, alpha(s->aa));
> +
> + return 0;
> +}
> +
> +static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
> +{
> + const int winlen = s->winlen;
> + const int tabsize = s->tabsize;
> + int i, cires = s->cur_ires;
> + float *nires;
> +
> + if (fs <= 0)
> + return;
> +
> + process_param(lbc, param, fs, 0);
> + for (i = 0; i < winlen; i++)
> + s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
> + for (; i < tabsize; i++)
> + s->irest[i] = 0;
> +
> + av_rdft_calc(s->rdft, s->irest);
> + nires = s->lires;
> + for (i = 0; i < tabsize; i++)
> + nires[i] = s->irest[i];
> +
> + process_param(rbc, param, fs, 1);
> + for (i = 0; i < winlen; i++)
> + s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
> + for (; i < tabsize; i++)
> + s->irest[i] = 0;
> +
> + av_rdft_calc(s->rdft, s->irest);
> + nires = s->rires;
> + for (i = 0; i < tabsize; i++)
> + nires[i] = s->irest[i];
> + s->chg_ires = cires == 1 ? 2 : 1;
> +}
> +
> +static int filter_frame(AVFilterLink *inlink, AVFrame *in)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + SuperEqualizerContext *s = ctx->priv;
> + AVFilterLink *outlink = ctx->outputs[0];
> + const float *ires = s->lires;
> + float *fsamples = s->fsamples;
> + int ch, i;
> +
> + AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
> + float *src, *dst, *ptr;
> +
> + if (!out) {
> + av_frame_free(&in);
> + return AVERROR(ENOMEM);
> + }
> +
> + for (ch = 0; ch < in->channels; ch++) {
> + ptr = (float *)out->extended_data[ch];
> + dst = (float *)s->out->extended_data[ch];
> + src = (float *)in->extended_data[ch];
> +
> + for (i = 0; i < s->winlen; i++)
> + fsamples[i] = src[i];
> + for (; i < s->tabsize; i++)
> + fsamples[i] = 0;
> +
> + av_rdft_calc(s->rdft, fsamples);
> +
> + fsamples[0] = ires[0] * fsamples[0];
> + fsamples[1] = ires[1] * fsamples[1];
> + for (i = 1; i < s->tabsize / 2; i++) {
> + float re, im;
> +
> + re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
> + im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
> +
> + fsamples[i*2 ] = re;
> + fsamples[i*2+1] = im;
> + }
> +
> + av_rdft_calc(s->irdft, fsamples);
> +
> + for (i = 0; i < s->winlen; i++)
> + dst[i] += fsamples[i] / s->tabsize * 2;
> + for (i = s->winlen; i < s->tabsize; i++)
> + dst[i] = fsamples[i] / s->tabsize * 2;
> + for (i = 0; i < s->winlen; i++)
> + ptr[i] = dst[i];
> + for (i = 0; i < s->winlen; i++)
> + dst[i] = dst[i+s->winlen];
> + }
> +
> + out->pts = in->pts;
> + av_frame_free(&in);
> +
> + return ff_filter_frame(outlink, out);
> +}
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> + SuperEqualizerContext *s = ctx->priv;
> +
> + return equ_init(s, 14);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterFormats *formats;
> + AVFilterChannelLayouts *layouts;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_FLTP,
> + AV_SAMPLE_FMT_NONE
> + };
> + int ret;
> +
> + layouts = ff_all_channel_counts();
> + if (!layouts)
> + return AVERROR(ENOMEM);
> + ret = ff_set_common_channel_layouts(ctx, layouts);
> + if (ret < 0)
> + return ret;
> +
> + formats = ff_make_format_list(sample_fmts);
> + if ((ret = ff_set_common_formats(ctx, formats)) < 0)
> + return ret;
> +
> + formats = ff_all_samplerates();
> + return ff_set_common_samplerates(ctx, formats);
> +}
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + SuperEqualizerContext *s = ctx->priv;
> +
> + inlink->partial_buf_size =
> + inlink->min_samples =
> + inlink->max_samples = s->winlen;
> +
> + s->out = ff_get_audio_buffer(inlink, s->tabsize);
> + if (!s->out)
> + return AVERROR(ENOMEM);
> +
> + return 0;
> +}
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + SuperEqualizerContext *s = ctx->priv;
> +
> + make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
> +
> + return 0;
> +}
> +
> +static av_cold void uninit(AVFilterContext *ctx)
> +{
> + SuperEqualizerContext *s = ctx->priv;
> +
> + av_rdft_end(s->rdft);
> + av_rdft_end(s->irdft);
> +}
> +
> +static const AVFilterPad superequalizer_inputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .filter_frame = filter_frame,
> + .config_props = config_input,
> + },
> + { NULL }
> +};
> +
> +static const AVFilterPad superequalizer_outputs[] = {
> + {
> + .name = "default",
> + .type = AVMEDIA_TYPE_AUDIO,
> + .config_props = config_output,
> + },
> + { NULL }
> +};
> +
> +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +#define OFFSET(x) offsetof(SuperEqualizerContext, x)
> +
> +static const AVOption superequalizer_options[] = {
> + { "1b", "set 1st band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "2b", "set 2nd band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "3b", "set 3rd band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "4b", "set 4th band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "5b", "set 5th band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "6b", "set 6th band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "7b", "set 7th band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "8b", "set 8th band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "9b", "set 9th band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "10b", "set 10nd band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "11b", "set 11nd band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "12b", "set 12nd band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "13b", "set 13nd band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "14b", "set 14th band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "15b", "set 15th band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "16b", "set 16th band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "17b", "set 17th band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { "18b", "set 18th band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(superequalizer);
> +
> +AVFilter ff_af_superequalizer = {
> + .name = "superequalizer",
> + .description = NULL_IF_CONFIG_SMALL("Apply 18-th band equalization filter."),
> + .priv_size = sizeof(SuperEqualizerContext),
> + .priv_class = &superequalizer_class,
> + .init = init,
> + .query_formats = query_formats,
> + .uninit = uninit,
> + .inputs = superequalizer_inputs,
> + .outputs = superequalizer_outputs,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index 94f7cf3..bd81091 100644
> --- a/libavfilter/allfilters.c
> +++ b/libavfilter/allfilters.c
> @@ -122,6 +122,7 @@ static void register_all(void)
> REGISTER_FILTER(SOFALIZER, sofalizer, af);
> REGISTER_FILTER(STEREOTOOLS, stereotools, af);
> REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
> + REGISTER_FILTER(SUPEREQUALIZER, superequalizer, af);
> REGISTER_FILTER(SURROUND, surround, af);
> REGISTER_FILTER(TREBLE, treble, af);
> REGISTER_FILTER(TREMOLO, tremolo, af);
>
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