[FFmpeg-devel] [PATCH] avfilter: add superequalizer filter
Paul B Mahol
onemda at gmail.com
Sat Jun 17 19:00:01 EEST 2017
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];
+ 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);
+}
+
+static float hn_imp(int n)
+{
+ return n == 0 ? 1.0 : 0.0;
+}
+
+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;
+}
+
+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));
+
+ 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);
--
2.9.3
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