@@ -1003,6 +1003,60 @@ aeval=val(0)|-val(1)
@end example
@end itemize
+@section aexciter
+
+An exciter is used to produce high sound that is not present in the
+original signal. This is done by creating harmonic distortions of the
+signal which are restricted in range and added to the original signal.
+An Exciter raises the upper end of an audio signal without simply raising
+the higher frequencies like an equalizer would do to create a more
+"crisp" or "brilliant" sound.
+
+The filter accepts the following options:
+
+@table @option
+@item level_in
+Set input level prior processing of signal.
+Allowed range is from 0 to 64.
+Default value is 1.
+
+@item level_out
+Set output level after processing of signal.
+Allowed range is from 0 to 64.
+Default value is 1.
+
+@item amount
+Set the amount of harmonics added to original signal.
+Allowed range is from 0 to 64.
+Default is 1.
+
+@item drive
+Set them amount of newly created harmonics.
+
+@item blend
+Set the octave of newly created harmonics.
+Allowed range is from -10 to 10.
+Default value is 0.
+
+@item freq
+Set the lower frequency limit of producing harmonics in Hz.
+Allowed range is from 2000 to 12000 Hz.
+Default is 7500 Hz.
+
+@item ceil
+Set the upper frequency limit of producing harmonics.
+Allowed range is from 9999 to 20000 Hz.
+If value is lower than 10000 Hz no limit is applied.
+
+@item listen
+Mute the original signal and output only added harmonics.
+By default is disabled.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@anchor{afade}
@section afade
@@ -46,6 +46,7 @@ OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
+OBJS-$(CONFIG_AEXCITER_FILTER) += af_aexciter.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFFTDN_FILTER) += af_afftdn.o
OBJS-$(CONFIG_AFFTFILT_FILTER) += af_afftfilt.o
new file mode 100644
@@ -0,0 +1,317 @@
+/*
+ * Copyright (c) Markus Schmidt and Christian Holschuh
+ *
+ * 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 "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct ChannelParams {
+ double blend_old, drive_old;
+ double rdrive, rbdr, kpa, kpb, kna, knb, ap,
+ an, imr, kc, srct, sq, pwrq;
+ double prev_med, prev_out;
+
+ double hp[5], lp[5];
+ double hw[4][2], lw[2][2];
+} ChannelParams;
+
+typedef struct AExciterContext {
+ const AVClass *class;
+
+ double level_in;
+ double level_out;
+ double amount;
+ double drive;
+ double blend;
+ double freq;
+ double ceil;
+ int listen;
+
+ ChannelParams *cp;
+} AExciterContext;
+
+#define OFFSET(x) offsetof(AExciterContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption aexciter_options[] = {
+ { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
+ { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
+ { "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A },
+ { "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A },
+ { "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A },
+ { "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A },
+ { "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A },
+ { "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aexciter);
+
+static inline double M(double x)
+{
+ return (fabs(x) > 0.00000001) ? x : 0.0;
+}
+
+static inline double D(double x)
+{
+ x = fabs(x);
+
+ return (x > 0.00000001) ? sqrt(x) : 0.0;
+}
+
+static void set_params(ChannelParams *p,
+ double blend, double drive,
+ double srate, double freq,
+ double ceil)
+{
+ double a0, a1, a2, b0, b1, b2, w0, alpha;
+
+ p->rdrive = 12.0 / drive;
+ p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0;
+ p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0;
+ p->kpb = (2.0 - p->kpa) / 2.0;
+ p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0;
+ p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive);
+
+ p->srct = (0.1 * srate) / (0.1 * srate + 1.0);
+ p->sq = p->kc*p->kc + 1.0;
+ p->knb = -1.0 * p->rbdr / D(p->sq);
+ p->kna = 2.0 * p->kc * p->rbdr / D(p->sq);
+ p->an = p->rbdr*p->rbdr / p->sq;
+ p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0);
+ p->pwrq = 2.0 / (p->imr + 1.0);
+
+ w0 = 2 * M_PI * freq / srate;
+ alpha = sin(w0) / (2. * 0.707);
+ a0 = 1 + alpha;
+ a1 = -2 * cos(w0);
+ a2 = 1 - alpha;
+ b0 = (1 + cos(w0)) / 2;
+ b1 = -(1 + cos(w0));
+ b2 = (1 + cos(w0)) / 2;
+
+ p->hp[0] =-a1 / a0;
+ p->hp[1] =-a2 / a0;
+ p->hp[2] = b0 / a0;
+ p->hp[3] = b1 / a0;
+ p->hp[4] = b2 / a0;
+
+ w0 = 2 * M_PI * ceil / srate;
+ alpha = sin(w0) / (2. * 0.707);
+ a0 = 1 + alpha;
+ a1 = -2 * cos(w0);
+ a2 = 1 - alpha;
+ b0 = (1 - cos(w0)) / 2;
+ b1 = 1 - cos(w0);
+ b2 = (1 - cos(w0)) / 2;
+
+ p->lp[0] =-a1 / a0;
+ p->lp[1] =-a2 / a0;
+ p->lp[2] = b0 / a0;
+ p->lp[3] = b1 / a0;
+ p->lp[4] = b2 / a0;
+}
+
+static double bprocess(double in, const double *const c,
+ double *w1, double *w2)
+{
+ double out = c[2] * in + *w1;
+
+ *w1 = c[3] * in + *w2 + c[0] * out;
+ *w2 = c[4] * in + c[1] * out;
+
+ return out;
+}
+
+static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
+{
+ double proc = in, med;
+
+ proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]);
+ proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]);
+
+ if (proc >= 0.0) {
+ med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq;
+ } else {
+ med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0;
+ }
+
+ proc = p->srct * (med - p->prev_med + p->prev_out);
+ p->prev_med = M(med);
+ p->prev_out = M(proc);
+
+ proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]);
+ proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]);
+
+ if (s->ceil >= 10000.) {
+ proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]);
+ proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]);
+ }
+
+ return proc;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AExciterContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out;
+ const double *src = (const double *)in->data[0];
+ const double level_in = s->level_in;
+ const double level_out = s->level_out;
+ const double amount = s->amount;
+ const double listen = 1.0 - s->listen;
+ double *dst;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ out = ff_get_audio_buffer(inlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+ }
+
+ dst = (double *)out->data[0];
+ for (int n = 0; n < in->nb_samples; n++) {
+ for (int c = 0; c < inlink->channels; c++) {
+ double sample = src[c] * level_in;
+
+ sample = distortion_process(s, &s->cp[c], sample);
+ sample = sample * amount + listen * src[c];
+
+ sample *= level_out;
+ if (ctx->is_disabled)
+ dst[c] = src[c];
+ else
+ dst[c] = sample;
+ }
+
+ src += inlink->channels;
+ dst += inlink->channels;
+ }
+
+ if (in != out)
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL,
+ 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 (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AExciterContext *s = ctx->priv;
+
+ av_freep(&s->cp);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AExciterContext *s = ctx->priv;
+
+ if (!s->cp)
+ s->cp = av_calloc(inlink->channels, sizeof(*s->cp));
+ if (!s->cp)
+ return AVERROR(ENOMEM);
+
+ for (int i = 0; i < inlink->channels; i++)
+ set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate,
+ s->freq, s->ceil);
+
+ return 0;
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+ char *res, int res_len, int flags)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ int ret;
+
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
+
+ return config_input(inlink);
+}
+
+static const AVFilterPad avfilter_af_aexciter_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_aexciter_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_aexciter = {
+ .name = "aexciter",
+ .description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."),
+ .priv_size = sizeof(AExciterContext),
+ .priv_class = &aexciter_class,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = avfilter_af_aexciter_inputs,
+ .outputs = avfilter_af_aexciter_outputs,
+ .process_command = process_command,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+};
@@ -39,6 +39,7 @@ extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
extern AVFilter ff_af_aemphasis;
extern AVFilter ff_af_aeval;
+extern AVFilter ff_af_aexciter;
extern AVFilter ff_af_afade;
extern AVFilter ff_af_afftdn;
extern AVFilter ff_af_afftfilt;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 54 +++++++ libavfilter/Makefile | 1 + libavfilter/af_aexciter.c | 317 ++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 373 insertions(+) create mode 100644 libavfilter/af_aexciter.c