[FFmpeg-cvslog] avfilter/af_adynamicequalizer: add precision option
Paul B Mahol
git at videolan.org
Sun Apr 30 13:41:00 EEST 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Apr 29 10:40:18 2023 +0200| [f247a3d82d9521394d71c4f95f87789af71de0cc] | committer: Paul B Mahol
avfilter/af_adynamicequalizer: add precision option
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=f247a3d82d9521394d71c4f95f87789af71de0cc
---
doc/filters.texi | 14 ++
libavfilter/adynamicequalizer_template.c | 270 +++++++++++++++++++++++++++++++
libavfilter/af_adynamicequalizer.c | 270 +++++++------------------------
3 files changed, 342 insertions(+), 212 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 50e1682144..f89b1d0b52 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1107,6 +1107,20 @@ Stop picking threshold value.
@item on
Start picking threshold value.
@end table
+
+ at item precision
+Set which precision to use when processing samples.
+
+ at table @option
+ at item auto
+Auto pick internal sample format depending on other filters.
+
+ at item float
+Always use single-floating point precision sample format.
+
+ at item double
+Always use double-floating point precision sample format.
+ at end table
@end table
@subsection Commands
diff --git a/libavfilter/adynamicequalizer_template.c b/libavfilter/adynamicequalizer_template.c
new file mode 100644
index 0000000000..ab180a9b9d
--- /dev/null
+++ b/libavfilter/adynamicequalizer_template.c
@@ -0,0 +1,270 @@
+/*
+ * 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
+ */
+
+#undef ftype
+#undef SQRT
+#undef TAN
+#undef ONE
+#undef TWO
+#undef ZERO
+#undef FMAX
+#undef FMIN
+#undef CLIP
+#undef SAMPLE_FORMAT
+#undef FABS
+#if DEPTH == 32
+#define SAMPLE_FORMAT float
+#define SQRT sqrtf
+#define TAN tanf
+#define ONE 1.f
+#define TWO 2.f
+#define ZERO 0.f
+#define FMIN fminf
+#define FMAX fmaxf
+#define CLIP av_clipf
+#define FABS fabsf
+#define ftype float
+#else
+#define SAMPLE_FORMAT double
+#define SQRT sqrt
+#define TAN tan
+#define ONE 1.0
+#define TWO 2.0
+#define ZERO 0.0
+#define FMIN fmin
+#define FMAX fmax
+#define CLIP av_clipd
+#define FABS fabs
+#define ftype double
+#endif
+
+#define fn3(a,b) a##_##b
+#define fn2(a,b) fn3(a,b)
+#define fn(a) fn2(a, SAMPLE_FORMAT)
+
+static ftype fn(get_svf)(ftype in, const ftype *m, const ftype *a, ftype *b)
+{
+ const ftype v0 = in;
+ const ftype v3 = v0 - b[1];
+ const ftype v1 = a[0] * b[0] + a[1] * v3;
+ const ftype v2 = b[1] + a[1] * b[0] + a[2] * v3;
+
+ b[0] = TWO * v1 - b[0];
+ b[1] = TWO * v2 - b[1];
+
+ return m[0] * v0 + m[1] * v1 + m[2] * v2;
+}
+
+static int fn(filter_prepare)(AVFilterContext *ctx)
+{
+ AudioDynamicEqualizerContext *s = ctx->priv;
+ const ftype sample_rate = ctx->inputs[0]->sample_rate;
+ const ftype dfrequency = FMIN(s->dfrequency, sample_rate * 0.5);
+ const ftype dg = TAN(M_PI * dfrequency / sample_rate);
+ const ftype dqfactor = s->dqfactor;
+ const int dftype = s->dftype;
+ ftype *da = fn(s->da);
+ ftype *dm = fn(s->dm);
+ ftype k;
+
+ s->attack_coef = get_coef(s->attack, sample_rate);
+ s->release_coef = get_coef(s->release, sample_rate);
+
+ switch (dftype) {
+ case 0:
+ k = ONE / dqfactor;
+
+ da[0] = ONE / (ONE + dg * (dg + k));
+ da[1] = dg * da[0];
+ da[2] = dg * da[1];
+
+ dm[0] = ZERO;
+ dm[1] = k;
+ dm[2] = ZERO;
+ break;
+ case 1:
+ k = ONE / dqfactor;
+
+ da[0] = ONE / (ONE + dg * (dg + k));
+ da[1] = dg * da[0];
+ da[2] = dg * da[1];
+
+ dm[0] = ZERO;
+ dm[1] = ZERO;
+ dm[2] = ONE;
+ break;
+ case 2:
+ k = ONE / dqfactor;
+
+ da[0] = ONE / (ONE + dg * (dg + k));
+ da[1] = dg * da[0];
+ da[2] = dg * da[1];
+
+ dm[0] = ZERO;
+ dm[1] = -k;
+ dm[2] = -ONE;
+ break;
+ case 3:
+ k = ONE / dqfactor;
+
+ da[0] = ONE / (ONE + dg * (dg + k));
+ da[1] = dg * da[0];
+ da[2] = dg * da[1];
+
+ dm[0] = ZERO;
+ dm[1] = -k;
+ dm[2] = -TWO;
+ break;
+ }
+
+ return 0;
+}
+
+static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AudioDynamicEqualizerContext *s = ctx->priv;
+ ThreadData *td = arg;
+ AVFrame *in = td->in;
+ AVFrame *out = td->out;
+ const ftype sample_rate = in->sample_rate;
+ const ftype makeup = s->makeup;
+ const ftype ratio = s->ratio;
+ const ftype range = s->range;
+ const ftype tfrequency = FMIN(s->tfrequency, sample_rate * 0.5);
+ const ftype release = s->release_coef;
+ const ftype irelease = ONE - release;
+ const ftype attack = s->attack_coef;
+ const ftype iattack = ONE - attack;
+ const ftype tqfactor = s->tqfactor;
+ const ftype itqfactor = ONE / tqfactor;
+ const ftype fg = TAN(M_PI * tfrequency / sample_rate);
+ const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
+ const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
+ const int detection = s->detection;
+ const int direction = s->direction;
+ const int tftype = s->tftype;
+ const int mode = s->mode;
+ const ftype *da = fn(s->da);
+ const ftype *dm = fn(s->dm);
+
+ for (int ch = start; ch < end; ch++) {
+ const ftype *src = (const ftype *)in->extended_data[ch];
+ ftype *dst = (ftype *)out->extended_data[ch];
+ ftype *state = (ftype *)s->state->extended_data[ch];
+ const ftype threshold = detection == 0 ? state[5] : s->threshold;
+
+ if (detection < 0)
+ state[5] = threshold;
+
+ for (int n = 0; n < out->nb_samples; n++) {
+ ftype detect, gain, v, listen;
+ ftype fa[3], fm[3];
+ ftype k, g;
+
+ detect = listen = fn(get_svf)(src[n], dm, da, state);
+ detect = FABS(detect);
+
+ if (detection > 0)
+ state[5] = FMAX(state[5], detect);
+
+ if (direction == 0) {
+ if (detect < threshold) {
+ if (mode == 0)
+ detect = ONE / CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
+ else
+ detect = CLIP(ONE + makeup + (threshold - detect) * ratio, ONE, range);
+ } else {
+ detect = ONE;
+ }
+ } else {
+ if (detect > threshold) {
+ if (mode == 0)
+ detect = ONE / CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
+ else
+ detect = CLIP(ONE + makeup + (detect - threshold) * ratio, ONE, range);
+ } else {
+ detect = ONE;
+ }
+ }
+
+ if (direction == 0) {
+ if (detect > state[4]) {
+ detect = iattack * detect + attack * state[4];
+ } else {
+ detect = irelease * detect + release * state[4];
+ }
+ } else {
+ if (detect < state[4]) {
+ detect = iattack * detect + attack * state[4];
+ } else {
+ detect = irelease * detect + release * state[4];
+ }
+ }
+
+ if (state[4] != detect || n == 0) {
+ state[4] = gain = detect;
+
+ switch (tftype) {
+ case 0:
+ k = ONE / (tqfactor * gain);
+
+ fa[0] = ONE / (ONE + fg * (fg + k));
+ fa[1] = fg * fa[0];
+ fa[2] = fg * fa[1];
+
+ fm[0] = ONE;
+ fm[1] = k * (gain * gain - ONE);
+ fm[2] = ZERO;
+ break;
+ case 1:
+ k = itqfactor;
+ g = fg / SQRT(gain);
+
+ fa[0] = ONE / (ONE + g * (g + k));
+ fa[1] = g * fa[0];
+ fa[2] = g * fa[1];
+
+ fm[0] = ONE;
+ fm[1] = k * (gain - ONE);
+ fm[2] = gain * gain - ONE;
+ break;
+ case 2:
+ k = itqfactor;
+ g = fg / SQRT(gain);
+
+ fa[0] = ONE / (ONE + g * (g + k));
+ fa[1] = g * fa[0];
+ fa[2] = g * fa[1];
+
+ fm[0] = gain * gain;
+ fm[1] = k * (ONE - gain) * gain;
+ fm[2] = ONE - gain * gain;
+ break;
+ }
+ }
+
+ v = fn(get_svf)(src[n], fm, fa, &state[2]);
+ v = mode == -1 ? listen : v;
+ dst[n] = ctx->is_disabled ? src[n] : v;
+ }
+ }
+
+ return 0;
+}
+
+
diff --git a/libavfilter/af_adynamicequalizer.c b/libavfilter/af_adynamicequalizer.c
index e741b55ead..a3aeee91c5 100644
--- a/libavfilter/af_adynamicequalizer.c
+++ b/libavfilter/af_adynamicequalizer.c
@@ -43,242 +43,82 @@ typedef struct AudioDynamicEqualizerContext {
int detection;
int tftype;
int dftype;
+ int precision;
+ int format;
+
+ int (*filter_prepare)(AVFilterContext *ctx);
+ int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
+
+ double da_double[3], dm_double[3];
+ float da_float[3], dm_float[3];
- double da[3], dm[3];
AVFrame *state;
} AudioDynamicEqualizerContext;
-static int config_input(AVFilterLink *inlink)
+static int query_formats(AVFilterContext *ctx)
{
- AVFilterContext *ctx = inlink->dst;
AudioDynamicEqualizerContext *s = ctx->priv;
+ static const enum AVSampleFormat sample_fmts[3][3] = {
+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
+ { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
+ };
+ int ret;
- s->state = ff_get_audio_buffer(inlink, 8);
- if (!s->state)
- return AVERROR(ENOMEM);
-
- for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
- double *state = (double *)s->state->extended_data[ch];
+ if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
+ return ret;
- state[4] = 1.;
- }
+ if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
+ return ret;
- return 0;
+ return ff_set_common_all_samplerates(ctx);
}
-static double get_svf(double in, const double *m, const double *a, double *b)
+static double get_coef(double x, double sr)
{
- const double v0 = in;
- const double v3 = v0 - b[1];
- const double v1 = a[0] * b[0] + a[1] * v3;
- const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
-
- b[0] = 2. * v1 - b[0];
- b[1] = 2. * v2 - b[1];
-
- return m[0] * v0 + m[1] * v1 + m[2] * v2;
+ return exp(-1000. / (x * sr));
}
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
-static double get_coef(double x, double sr)
-{
- return exp(-1000. / (x * sr));
-}
+#define DEPTH 32
+#include "adynamicequalizer_template.c"
-static int filter_prepare(AVFilterContext *ctx)
+#undef DEPTH
+#define DEPTH 64
+#include "adynamicequalizer_template.c"
+
+static int config_input(AVFilterLink *inlink)
{
+ AVFilterContext *ctx = inlink->dst;
AudioDynamicEqualizerContext *s = ctx->priv;
- const double sample_rate = ctx->inputs[0]->sample_rate;
- const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
- const double dg = tan(M_PI * dfrequency / sample_rate);
- const double dqfactor = s->dqfactor;
- const int dftype = s->dftype;
- double *da = s->da;
- double *dm = s->dm;
- double k;
-
- s->attack_coef = get_coef(s->attack, sample_rate);
- s->release_coef = get_coef(s->release, sample_rate);
-
- switch (dftype) {
- case 0:
- k = 1. / dqfactor;
-
- da[0] = 1. / (1. + dg * (dg + k));
- da[1] = dg * da[0];
- da[2] = dg * da[1];
-
- dm[0] = 0.;
- dm[1] = k;
- dm[2] = 0.;
- break;
- case 1:
- k = 1. / dqfactor;
-
- da[0] = 1. / (1. + dg * (dg + k));
- da[1] = dg * da[0];
- da[2] = dg * da[1];
-
- dm[0] = 0.;
- dm[1] = 0.;
- dm[2] = 1.;
- break;
- case 2:
- k = 1. / dqfactor;
- da[0] = 1. / (1. + dg * (dg + k));
- da[1] = dg * da[0];
- da[2] = dg * da[1];
-
- dm[0] = 0.;
- dm[1] = -k;
- dm[2] = -1.;
- break;
- case 3:
- k = 1. / dqfactor;
+ s->format = inlink->format;
+ s->state = ff_get_audio_buffer(inlink, 8);
+ if (!s->state)
+ return AVERROR(ENOMEM);
- da[0] = 1. / (1. + dg * (dg + k));
- da[1] = dg * da[0];
- da[2] = dg * da[1];
+ switch (s->format) {
+ case AV_SAMPLE_FMT_DBLP:
+ for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
+ double *state = (double *)s->state->extended_data[ch];
- dm[0] = 0.;
- dm[1] = -k;
- dm[2] = -2.;
+ state[4] = 1.;
+ }
+ s->filter_prepare = filter_prepare_double;
+ s->filter_channels = filter_channels_double;
break;
- }
-
- return 0;
-}
+ case AV_SAMPLE_FMT_FLTP:
+ for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
+ float *state = (float *)s->state->extended_data[ch];
-static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
-{
- AudioDynamicEqualizerContext *s = ctx->priv;
- ThreadData *td = arg;
- AVFrame *in = td->in;
- AVFrame *out = td->out;
- const double sample_rate = in->sample_rate;
- const double makeup = s->makeup;
- const double ratio = s->ratio;
- const double range = s->range;
- const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
- const double release = s->release_coef;
- const double irelease = 1. - release;
- const double attack = s->attack_coef;
- const double iattack = 1. - attack;
- const double tqfactor = s->tqfactor;
- const double fg = tan(M_PI * tfrequency / sample_rate);
- const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
- const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
- const int detection = s->detection;
- const int direction = s->direction;
- const int tftype = s->tftype;
- const int mode = s->mode;
- const double *da = s->da;
- const double *dm = s->dm;
-
- for (int ch = start; ch < end; ch++) {
- const double *src = (const double *)in->extended_data[ch];
- double *dst = (double *)out->extended_data[ch];
- double *state = (double *)s->state->extended_data[ch];
- const double threshold = detection == 0 ? state[5] : s->threshold;
-
- if (detection < 0)
- state[5] = threshold;
-
- for (int n = 0; n < out->nb_samples; n++) {
- double detect, gain, v, listen;
- double fa[3], fm[3];
- double k, g;
-
- detect = listen = get_svf(src[n], dm, da, state);
- detect = fabs(detect);
-
- if (detection > 0)
- state[5] = fmax(state[5], detect);
-
- if (direction == 0) {
- if (detect < threshold) {
- if (mode == 0)
- detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
- else
- detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
- } else {
- detect = 1.;
- }
- } else {
- if (detect > threshold) {
- if (mode == 0)
- detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
- else
- detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
- } else {
- detect = 1.;
- }
- }
-
- if (direction == 0) {
- if (detect > state[4]) {
- detect = iattack * detect + attack * state[4];
- } else {
- detect = irelease * detect + release * state[4];
- }
- } else {
- if (detect < state[4]) {
- detect = iattack * detect + attack * state[4];
- } else {
- detect = irelease * detect + release * state[4];
- }
- }
-
- if (state[4] != detect || n == 0) {
- state[4] = gain = detect;
-
- switch (tftype) {
- case 0:
- k = 1. / (tqfactor * gain);
-
- fa[0] = 1. / (1. + fg * (fg + k));
- fa[1] = fg * fa[0];
- fa[2] = fg * fa[1];
-
- fm[0] = 1.;
- fm[1] = k * (gain * gain - 1.);
- fm[2] = 0.;
- break;
- case 1:
- k = 1. / tqfactor;
- g = fg / sqrt(gain);
-
- fa[0] = 1. / (1. + g * (g + k));
- fa[1] = g * fa[0];
- fa[2] = g * fa[1];
-
- fm[0] = 1.;
- fm[1] = k * (gain - 1.);
- fm[2] = gain * gain - 1.;
- break;
- case 2:
- k = 1. / tqfactor;
- g = fg / sqrt(gain);
-
- fa[0] = 1. / (1. + g * (g + k));
- fa[1] = g * fa[0];
- fa[2] = g * fa[1];
-
- fm[0] = gain * gain;
- fm[1] = k * (1. - gain) * gain;
- fm[2] = 1. - gain * gain;
- break;
- }
- }
-
- v = get_svf(src[n], fm, fa, &state[2]);
- v = mode == -1 ? listen : v;
- dst[n] = ctx->is_disabled ? src[n] : v;
+ state[4] = 1.;
}
+ s->filter_prepare = filter_prepare_float;
+ s->filter_channels = filter_channels_float;
+ break;
}
return 0;
@@ -288,6 +128,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
+ AudioDynamicEqualizerContext *s = ctx->priv;
ThreadData td;
AVFrame *out;
@@ -304,8 +145,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
td.in = in;
td.out = out;
- filter_prepare(ctx);
- ff_filter_execute(ctx, filter_channels, &td, NULL,
+ s->filter_prepare(ctx);
+ ff_filter_execute(ctx, s->filter_channels, &td, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
if (out != in)
@@ -321,6 +162,7 @@ static av_cold void uninit(AVFilterContext *ctx)
}
#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption adynamicequalizer_options[] = {
@@ -354,6 +196,10 @@ static const AVOption adynamicequalizer_options[] = {
{ "disabled", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "auto" },
{ "off", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "auto" },
{ "on", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "auto" },
+ { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, "precision" },
+ { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision" },
+ { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision" },
+ { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision" },
{ NULL }
};
@@ -383,7 +229,7 @@ const AVFilter ff_af_adynamicequalizer = {
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
- FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
+ FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
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