Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 49 +++++
libavfilter/Makefile | 1 +
libavfilter/af_speechnorm.c | 422 ++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 473 insertions(+)
create mode 100644 libavfilter/af_speechnorm.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 8380f6cac2..7db43dd34c 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -5276,6 +5276,55 @@ and also with custom gain:
@end example
@end itemize
+@section speechnorm
+Speech Normalizer.
+
+This filter expands or compress each half-cycle of audio samples
+(local set of samples all above or all bellow zero) depending on threshold
value,
+so audio reaches target peak value under conditions controlled by bellow
options.
+
+The filter accepts the following options:
+
+@table @option
+@item peak, p
+Set the expansion target peak value. This specifies the highest allowed
absolute amplitude
+level for the normalized audio input. Default value is 0.95. Allowed range is
from 0.0 to 1.0.
+
+@item expansion, e
+Set the maximum expansion factor. Allowed range is from 1.0 to 10.0. Default
value is 2.0.
+This option controls maximum local half-cycle of samples expansion. The
maximum expansion
+would be such that local peak value reach target peak value but never to
surpass it and that
+ratio between new and previous peak value does not surpass this option value.
+
+@item compression, c
+Set the maximum compression factor. Allowed range is from 1.0 to 10.0. Default
value is 2.0.
+This option controls maximum local half-cycle of samples compression. This
option is used
+only if @option{threshold} option is set to value greater than 0.0, than in
such cases
+when local peak is lower or same as value set by @option{threshold} all
samples belonging to
+that peak's half-cycle would be compressed by current compression factor.
+
+@item threshold, t
+Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to
1.0.
+This option specify which half-cycles of samples will be compressed and which
will be expanded.
+Any half-cycle samples with their local peak value bellow or same as this
option value will be
+compressed by current compression factor, otherwise, if greater than threshold
value it will be
+expanded with expansion factor so that it could reach peak target value but
never surpass it.
+
+@item raise, r
+Set the expansion raising amount per each half-cycle of samples. Default value
is 0.001.
+Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is
raised per
+each new half-cycle until it reaches @option{expansion} value.
+
+@item fall, f
+Set the compression raising amount per each half-cycle of samples. Default
value is 0.001.
+Allowed range is from 0.0 to 1.0. This controls how fast compression factor is
raised per
+each new half-cycle until it reaches @option{compression} value.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section stereotools
This filter has some handy utilities to manage stereo signals, for converting
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 0c2a5d1cf4..36f3d2d0e4 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -138,6 +138,7 @@ OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
+OBJS-$(CONFIG_SPEECHNORM_FILTER) += af_speechnorm.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o
diff --git a/libavfilter/af_speechnorm.c b/libavfilter/af_speechnorm.c
new file mode 100644
index 0000000000..ef06c72fee
--- /dev/null
+++ b/libavfilter/af_speechnorm.c
@@ -0,0 +1,422 @@
+/*
+ * Speech Normalizer
+ * Copyright (c) 2020 Paul B Mahol
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Speech Normalizer
+ */
+
+#include <float.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+
+#define FF_BUFQUEUE_SIZE (1024)
+#include "bufferqueue.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "filters.h"
+#include "internal.h"
+
+#define MAX_ITEMS 882000
+
+typedef struct PeriodItem {
+ int size;
+ int type;
+ double max_peak;
+} PeriodItem;
+
+typedef struct ChannelContext {
+ int state;
+ PeriodItem pi[MAX_ITEMS];
+ double gain_state;
+ int pi_start;
+ int pi_end;
+} ChannelContext;
+
+typedef struct SpeechNormalizerContext {
+ const AVClass *class;
+
+ double peak_value;
+ double max_expansion;
+ double max_compression;
+ double threshold_value;
+ double raise_amount;
+ double fall_amount;
+ int channels;
+
+ ChannelContext *cc;
+
+ int max_period;
+ int eof;
+ int64_t pts;
+
+ struct FFBufQueue queue;
+} SpeechNormalizerContext;
+
+#define OFFSET(x) offsetof(SpeechNormalizerContext, x)
+#define FLAGS
AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption speechnorm_options[] = {
+ { "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE,
{.dbl=0.95}, 0.0, 1.0, FLAGS },
+ { "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE,
{.dbl=0.95}, 0.0, 1.0, FLAGS },
+ { "expansion", "set the max expansion factor", OFFSET(max_expansion),
AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 10.0, FLAGS },
+ { "e", "set the max expansion factor", OFFSET(max_expansion),
AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 10.0, FLAGS },
+ { "compresion", "set the max compression factor", OFFSET(max_compression),
AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 10.0, FLAGS },
+ { "c", "set the max compression factor", OFFSET(max_compression),
AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 10.0, FLAGS },
+ { "threshold", "set the threshold value", OFFSET(threshold_value),
AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
+ { "t", "set the threshold value", OFFSET(threshold_value),
AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
+ { "raise", "set the expansion raising amount", OFFSET(raise_amount),
AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
+ { "r", "set the expansion raising amount", OFFSET(raise_amount),
AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
+ { "fall", "set the compression raising amount", OFFSET(fall_amount),
AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
+ { "f", "set the compression raising amount", OFFSET(fall_amount),
AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(speechnorm);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBLP,
+ 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 int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SpeechNormalizerContext *s = ctx->priv;
+
+ s->max_period = inlink->sample_rate / (2 * 20);
+ s->channels = inlink->channels;
+
+ s->cc = av_calloc(inlink->channels, sizeof(*s->cc));
+ if (!s->cc)
+ return AVERROR(ENOMEM);
+
+ for (int ch = 0; ch < s->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+
+ cc->state = -1;
+ cc->gain_state = 1.;
+ }
+
+ return 0;
+}
+
+static int get_pi_samples(PeriodItem *pi, int start, int end, int mode)
+{
+ int sum;
+
+ if (mode && pi[start].type == 0)
+ return 0;
+
+ sum = pi[start].size;
+ av_assert0(sum >= 0);
+ while (start != end) {
+ start++;
+ if (start >= MAX_ITEMS)
+ start = 0;
+ if (mode && pi[start].type == 0)
+ break;
+ av_assert0(pi[start].size > 0);
+ sum += pi[start].size;
+ if (pi[start].type == 0)
+ break;
+ }
+
+ return sum;
+}
+
+static int consume_pi(PeriodItem *pi, int start, int end, int nb_samples)
+{
+ int sum;
+
+ sum = pi[start].size;
+ av_assert0(pi[start].size > 0);
+ while (sum < nb_samples) {
+ av_assert0(pi[start].type == 1);
+ av_assert0(start != end);
+ start++;
+ if (start >= MAX_ITEMS)
+ start = 0;
+ av_assert0(pi[start].size > 0);
+ sum += pi[start].size;
+ }
+
+ av_assert0(pi[start].size >= sum - nb_samples);
+ pi[start].size = sum - nb_samples;
+ av_assert0(pi[start].size >= 0);
+ if (pi[start].size == 0 && start != end) {
+ start++;
+ if (start >= MAX_ITEMS)
+ start = 0;
+ }
+
+ return start;
+}
+
+static int get_queued_samples(SpeechNormalizerContext *s)
+{
+ int sum = 0;
+
+ for (int i = 0; i < s->queue.available; i++) {
+ AVFrame *frame = ff_bufqueue_peek(&s->queue, i);
+ sum += frame->nb_samples;
+ }
+
+ return sum;
+}
+
+static int filter_frame(AVFilterContext *ctx)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFilterLink *inlink = ctx->inputs[0];
+ int min_pi_nb_samples;
+ AVFrame *in = NULL;
+ int ret;
+
+ for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) {
+ ret = ff_inlink_consume_frame(inlink, &in);
+ if (ret < 0)
+ return ret;
+ if (ret == 0)
+ break;
+
+ ff_bufqueue_add(ctx, &s->queue, in);
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+ const double *src = (const double *)in->extended_data[ch];
+ int n = 0;
+
+ if (cc->state < 0)
+ cc->state = src[0] >= 0.;
+
+ while (n < in->nb_samples) {
+ if (cc->state != (src[n] >= 0.) || cc->pi[cc->pi_end].size >
s->max_period) {
+ cc->state = src[n] >= 0.;
+ av_assert0(cc->pi[cc->pi_end].size > 0);
+ cc->pi[cc->pi_end].type = 1;
+ cc->pi_end++;
+ if (cc->pi_end >= MAX_ITEMS)
+ cc->pi_end = 0;
+ cc->pi[cc->pi_end].max_peak = DBL_MIN;
+ cc->pi[cc->pi_end].type = 0;
+ cc->pi[cc->pi_end].size = 0;
+ av_assert0(cc->pi_end != cc->pi_start);
+ }
+
+ if (src[n] >= 0.) {
+ while (src[n] >= 0.) {
+ cc->pi[cc->pi_end].max_peak =
FFMAX(cc->pi[cc->pi_end].max_peak, FFABS(src[n]));
+ cc->pi[cc->pi_end].size++;
+ n++;
+ if (n >= in->nb_samples)
+ break;
+ }
+ } else {
+ while (src[n] < 0.) {
+ cc->pi[cc->pi_end].max_peak =
FFMAX(cc->pi[cc->pi_end].max_peak, FFABS(src[n]));
+ cc->pi[cc->pi_end].size++;
+ n++;
+ if (n >= in->nb_samples)
+ break;
+ }
+ }
+ }
+ }
+ }
+
+ if (s->queue.available > 0) {
+ in = ff_bufqueue_peek(&s->queue, 0);
+ if (!in)
+ return 1;
+ } else {
+ return 1;
+ }
+
+ min_pi_nb_samples = get_pi_samples(s->cc[0].pi, s->cc[0].pi_start,
s->cc[0].pi_end, 1);
+ for (int ch = 1; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+ min_pi_nb_samples = FFMIN(min_pi_nb_samples, get_pi_samples(cc->pi,
cc->pi_start, cc->pi_end, 1));
+ }
+
+ if (min_pi_nb_samples >= in->nb_samples || s->eof) {
+ int nb_samples = get_queued_samples(s);
+
+ in = ff_bufqueue_get(&s->queue);
+
+ av_frame_make_writable(in);
+
+ nb_samples -= in->nb_samples;
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+ double *src = (double *)in->extended_data[ch];
+ int start = cc->pi_start;
+ int offset = 0;
+ double gain = 1.;
+
+ for (int n = 0; n < in->nb_samples; n++) {
+ if (n >= offset) {
+ int type = cc->pi[start].max_peak > s->threshold_value;
+
+ if (type)
+ gain = FFMIN(s->max_expansion, s->peak_value /
cc->pi[start].max_peak);
+ else
+ gain = 1. / s->max_compression;
+
+ av_assert0(cc->pi[start].size > 0);
+ offset += cc->pi[start++].size;
+ if (start >= MAX_ITEMS)
+ start = 0;
+
+ if (type)
+ cc->gain_state = FFMIN(gain, cc->gain_state +
s->raise_amount);
+ else
+ cc->gain_state = FFMAX(gain, cc->gain_state -
s->fall_amount);
+ }
+ src[n] *= cc->gain_state;
+ }
+ }
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+
+ cc->pi_start = consume_pi(cc->pi, cc->pi_start, cc->pi_end,
in->nb_samples);
+ }
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+ int pi_nb_samples = get_pi_samples(cc->pi, cc->pi_start,
cc->pi_end, 0);
+
+ if (nb_samples != pi_nb_samples) {
+ av_assert0(0);
+ }
+ }
+
+ s->pts = in->pts + in->nb_samples;
+
+ return ff_filter_frame(outlink, in);
+ }
+
+ return 1;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ SpeechNormalizerContext *s = ctx->priv;
+ int ret = 0, status;
+ int64_t pts;
+
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+ ret = filter_frame(ctx);
+ if (ret <= 0)
+ return ret;
+
+ if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+ if (status == AVERROR_EOF)
+ s->eof = 1;
+ }
+
+ if (s->eof && ff_inlink_queued_samples(inlink) == 0 &&
+ s->queue.available == 0) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
+ return 0;
+ }
+
+ if (!s->eof)
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ if (s->eof && s->queue.available > 0) {
+ ff_filter_set_ready(ctx, 10);
+ return 0;
+ }
+
+ return FFERROR_NOT_READY;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+
+ ff_bufqueue_discard_all(&s->queue);
+ av_freep(&s->cc);
+}
+
+static const AVFilterPad avfilter_af_speechnorm_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_speechnorm_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_speechnorm = {
+ .name = "speechnorm",
+ .description = NULL_IF_CONFIG_SMALL("Speech Normalizer."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(SpeechNormalizerContext),
+ .priv_class = &speechnorm_class,
+ .activate = activate,
+ .uninit = uninit,
+ .inputs = avfilter_af_speechnorm_inputs,
+ .outputs = avfilter_af_speechnorm_outputs,
+ .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 7796959dc7..fde535d50c 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -132,6 +132,7 @@ extern AVFilter ff_af_sidechaingate;
extern AVFilter ff_af_silencedetect;
extern AVFilter ff_af_silenceremove;
extern AVFilter ff_af_sofalizer;
+extern AVFilter ff_af_speechnorm;
extern AVFilter ff_af_stereotools;
extern AVFilter ff_af_stereowiden;
extern AVFilter ff_af_superequalizer;
--
2.17.1
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