summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rw-r--r--Changelog1
-rw-r--r--doc/filters.texi63
-rw-r--r--libavfilter/Makefile1
-rw-r--r--libavfilter/af_speechnorm.c579
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/version.h2
6 files changed, 646 insertions, 1 deletions
diff --git a/Changelog b/Changelog
index d8f151b02d..fcdabce2f2 100644
--- a/Changelog
+++ b/Changelog
@@ -45,6 +45,7 @@ version <next>:
- AMV muxer
- NVDEC AV1 hwaccel
- DXVA2/D3D11VA hardware accelerated AV1 decoding
+- speechnorm filter
version 4.3:
diff --git a/doc/filters.texi b/doc/filters.texi
index 8380f6cac2..8c9eccb00e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -5276,6 +5276,69 @@ and also with custom gain:
@end example
@end itemize
+@section speechnorm
+Speech Normalizer.
+
+This filter expands or compresses each half-cycle of audio samples
+(local set of samples all above or all below zero and between two nearest zero crossings) depending
+on threshold value, so audio reaches target peak value under conditions controlled by below 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 50.0. 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 reaches 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 50.0. 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, then 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 will 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 specifies which half-cycles of samples will be compressed and which will be expanded.
+Any half-cycle samples with their local peak value below or same as this option value will be
+compressed by current compression factor, otherwise, if greater than threshold value they 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.
+Setting this options too high may lead to distortions.
+
+@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.
+
+@item channels, h
+Specify which channels to filter, by default all available channels are filtered.
+
+@item invert, i
+Enable inverted filtering, by default is disabled. This inverts interpretation of @option{threshold}
+option. When enabled any half-cycle of samples with their local peak value below or same as
+@option{threshold} option will be expanded otherwise it will be compressed.
+
+@item link, l
+Link channels when calculating gain applied to each filtered channel sample, by default is disabled.
+When disabled each filtered channel gain calculation is independent, otherwise when this option
+is enabled the minimum of all possible gains for each filtered channel is used.
+@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 30e39b7f83..794a55ac3d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -137,6 +137,7 @@ 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..eb46cf2985
--- /dev/null
+++ b/libavfilter/af_speechnorm.c
@@ -0,0 +1,579 @@
+/*
+ * Copyright (c) 2020 Paul B Mahol
+ *
+ * Speech Normalizer
+ *
+ * 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
+#define MIN_PEAK (1. / 32768.)
+
+typedef struct PeriodItem {
+ int size;
+ int type;
+ double max_peak;
+} PeriodItem;
+
+typedef struct ChannelContext {
+ int state;
+ int bypass;
+ PeriodItem pi[MAX_ITEMS];
+ double gain_state;
+ double pi_max_peak;
+ int pi_start;
+ int pi_end;
+ int pi_size;
+} 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;
+ uint64_t channels;
+ int invert;
+ int link;
+
+ ChannelContext *cc;
+ double prev_gain;
+
+ int max_period;
+ int eof;
+ int64_t pts;
+
+ struct FFBufQueue queue;
+
+ void (*analyze_channel)(AVFilterContext *ctx, ChannelContext *cc,
+ const uint8_t *srcp, int nb_samples);
+ void (*filter_channels[2])(AVFilterContext *ctx,
+ AVFrame *in, int nb_samples);
+} 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, 50.0, FLAGS },
+ { "e", "set the max expansion factor", OFFSET(max_expansion), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
+ { "compression", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
+ { "c", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, 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 },
+ { "channels", "set channels to filter", OFFSET(channels), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
+ { "h", "set channels to filter", OFFSET(channels), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
+ { "invert", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
+ { "i", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
+ { "link", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
+ { "l", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, 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_FLTP, 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 get_pi_samples(PeriodItem *pi, int start, int end, int remain)
+{
+ int sum;
+
+ if (pi[start].type == 0)
+ return remain;
+
+ sum = remain;
+ while (start != end) {
+ start++;
+ if (start >= MAX_ITEMS)
+ start = 0;
+ if (pi[start].type == 0)
+ break;
+ av_assert0(pi[start].size > 0);
+ sum += pi[start].size;
+ }
+
+ return sum;
+}
+
+static int available_samples(AVFilterContext *ctx)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ int min_pi_nb_samples;
+
+ min_pi_nb_samples = get_pi_samples(s->cc[0].pi, s->cc[0].pi_start, s->cc[0].pi_end, s->cc[0].pi_size);
+ for (int ch = 1; ch < inlink->channels && min_pi_nb_samples > 0; 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, cc->pi_size));
+ }
+
+ return min_pi_nb_samples;
+}
+
+static void consume_pi(ChannelContext *cc, int nb_samples)
+{
+ if (cc->pi_size >= nb_samples) {
+ cc->pi_size -= nb_samples;
+ } else {
+ av_assert0(0);
+ }
+}
+
+static double next_gain(AVFilterContext *ctx, double pi_max_peak, int bypass, double state)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ const double expansion = FFMIN(s->max_expansion, s->peak_value / pi_max_peak);
+ const double compression = 1. / s->max_compression;
+ const int type = s->invert ? pi_max_peak <= s->threshold_value : pi_max_peak >= s->threshold_value;
+
+ if (bypass) {
+ return 1.;
+ } else if (type) {
+ return FFMIN(expansion, state + s->raise_amount);
+ } else {
+ return FFMIN(expansion, FFMAX(compression, state - s->fall_amount));
+ }
+}
+
+static void next_pi(AVFilterContext *ctx, ChannelContext *cc, int bypass)
+{
+ av_assert0(cc->pi_size >= 0);
+ if (cc->pi_size == 0) {
+ SpeechNormalizerContext *s = ctx->priv;
+ int start = cc->pi_start;
+
+ av_assert0(cc->pi[start].size > 0);
+ av_assert0(cc->pi[start].type > 0 || s->eof);
+ cc->pi_size = cc->pi[start].size;
+ cc->pi_max_peak = cc->pi[start].max_peak;
+ av_assert0(cc->pi_start != cc->pi_end || s->eof);
+ start++;
+ if (start >= MAX_ITEMS)
+ start = 0;
+ cc->pi_start = start;
+ cc->gain_state = next_gain(ctx, cc->pi_max_peak, bypass, cc->gain_state);
+ }
+}
+
+static double min_gain(AVFilterContext *ctx, ChannelContext *cc, int max_size)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ double min_gain = s->max_expansion;
+ double gain_state = cc->gain_state;
+ int size = cc->pi_size;
+ int idx = cc->pi_start;
+
+ min_gain = FFMIN(min_gain, gain_state);
+ while (size <= max_size) {
+ if (idx == cc->pi_end)
+ break;
+ gain_state = next_gain(ctx, cc->pi[idx].max_peak, 0, gain_state);
+ min_gain = FFMIN(min_gain, gain_state);
+ size += cc->pi[idx].size;
+ idx++;
+ if (idx >= MAX_ITEMS)
+ idx = 0;
+ }
+
+ return min_gain;
+}
+
+#define ANALYZE_CHANNEL(name, ptype, zero) \
+static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
+ const uint8_t *srcp, int nb_samples) \
+{ \
+ SpeechNormalizerContext *s = ctx->priv; \
+ const ptype *src = (const ptype *)srcp; \
+ int n = 0; \
+ \
+ if (cc->state < 0) \
+ cc->state = src[0] >= zero; \
+ \
+ while (n < nb_samples) { \
+ if ((cc->state != (src[n] >= zero)) || \
+ (cc->pi[cc->pi_end].size > s->max_period)) { \
+ double max_peak = cc->pi[cc->pi_end].max_peak; \
+ int state = cc->state; \
+ cc->state = src[n] >= zero; \
+ av_assert0(cc->pi[cc->pi_end].size > 0); \
+ if (cc->pi[cc->pi_end].max_peak >= MIN_PEAK || \
+ cc->pi[cc->pi_end].size > s->max_period) { \
+ cc->pi[cc->pi_end].type = 1; \
+ cc->pi_end++; \
+ if (cc->pi_end >= MAX_ITEMS) \
+ cc->pi_end = 0; \
+ if (cc->state != state) \
+ cc->pi[cc->pi_end].max_peak = DBL_MIN; \
+ else \
+ cc->pi[cc->pi_end].max_peak = max_peak; \
+ cc->pi[cc->pi_end].type = 0; \
+ cc->pi[cc->pi_end].size = 0; \
+ av_assert0(cc->pi_end != cc->pi_start); \
+ } \
+ } \
+ \
+ if (cc->state) { \
+ while (src[n] >= zero) { \
+ cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, src[n]); \
+ cc->pi[cc->pi_end].size++; \
+ n++; \
+ if (n >= nb_samples) \
+ break; \
+ } \
+ } else { \
+ while (src[n] < zero) { \
+ cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, -src[n]); \
+ cc->pi[cc->pi_end].size++; \
+ n++; \
+ if (n >= nb_samples) \
+ break; \
+ } \
+ } \
+ } \
+}
+
+ANALYZE_CHANNEL(dbl, double, 0.0)
+ANALYZE_CHANNEL(flt, float, 0.f)
+
+#define FILTER_CHANNELS(name, ptype) \
+static void filter_channels_## name (AVFilterContext *ctx, \
+ AVFrame *in, int nb_samples) \
+{ \
+ SpeechNormalizerContext *s = ctx->priv; \
+ AVFilterLink *inlink = ctx->inputs[0]; \
+ \
+ for (int ch = 0; ch < inlink->channels; ch++) { \
+ ChannelContext *cc = &s->cc[ch]; \
+ ptype *dst = (ptype *)in->extended_data[ch]; \
+ const int bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); \
+ int n = 0; \
+ \
+ while (n < nb_samples) { \
+ ptype gain; \
+ int size; \
+ \
+ next_pi(ctx, cc, bypass); \
+ size = FFMIN(nb_samples - n, cc->pi_size); \
+ av_assert0(size > 0); \
+ gain = cc->gain_state; \
+ consume_pi(cc, size); \
+ for (int i = n; i < n + size; i++) \
+ dst[i] *= gain; \
+ n += size; \
+ } \
+ } \
+}
+
+FILTER_CHANNELS(dbl, double)
+FILTER_CHANNELS(flt, float)
+
+static double lerp(double min, double max, double mix)
+{
+ return min + (max - min) * mix;
+}
+
+#define FILTER_LINK_CHANNELS(name, ptype) \
+static void filter_link_channels_## name (AVFilterContext *ctx, \
+ AVFrame *in, int nb_samples) \
+{ \
+ SpeechNormalizerContext *s = ctx->priv; \
+ AVFilterLink *inlink = ctx->inputs[0]; \
+ int n = 0; \
+ \
+ while (n < nb_samples) { \
+ int min_size = nb_samples - n; \
+ int max_size = 1; \
+ ptype gain = s->max_expansion; \
+ \
+ for (int ch = 0; ch < inlink->channels; ch++) { \
+ ChannelContext *cc = &s->cc[ch]; \
+ \
+ cc->bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); \
+ \
+ next_pi(ctx, cc, cc->bypass); \
+ min_size = FFMIN(min_size, cc->pi_size); \
+ max_size = FFMAX(max_size, cc->pi_size); \
+ } \
+ \
+ av_assert0(min_size > 0); \
+ for (int ch = 0; ch < inlink->channels; ch++) { \
+ ChannelContext *cc = &s->cc[ch]; \
+ \
+ if (cc->bypass) \
+ continue; \
+ gain = FFMIN(gain, min_gain(ctx, cc, max_size)); \
+ } \
+ \
+ for (int ch = 0; ch < inlink->channels; ch++) { \
+ ChannelContext *cc = &s->cc[ch]; \
+ ptype *dst = (ptype *)in->extended_data[ch]; \
+ \
+ consume_pi(cc, min_size); \
+ if (cc->bypass) \
+ continue; \
+ \
+ for (int i = n; i < n + min_size; i++) { \
+ ptype g = lerp(s->prev_gain, gain, (i - n) / (double)min_size); \
+ dst[i] *= g; \
+ } \
+ } \
+ \
+ s->prev_gain = gain; \
+ n += min_size; \
+ } \
+}
+
+FILTER_LINK_CHANNELS(dbl, double)
+FILTER_LINK_CHANNELS(flt, float)
+
+static int filter_frame(AVFilterContext *ctx)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFilterLink *inlink = ctx->inputs[0];
+ int ret;
+
+ while (s->queue.available > 0) {
+ int min_pi_nb_samples;
+ AVFrame *in;
+
+ in = ff_bufqueue_peek(&s->queue, 0);
+ if (!in)
+ break;
+
+ min_pi_nb_samples = available_samples(ctx);
+ if (min_pi_nb_samples < in->nb_samples && !s->eof)
+ break;
+
+ in = ff_bufqueue_get(&s->queue);
+
+ av_frame_make_writable(in);
+
+ s->filter_channels[s->link](ctx, in, in->nb_samples);
+
+ s->pts = in->pts + in->nb_samples;
+
+ return ff_filter_frame(outlink, in);
+ }
+
+ for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) {
+ AVFrame *in;
+
+ 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];
+
+ s->analyze_channel(ctx, cc, in->extended_data[ch], in->nb_samples);
+ }
+ }
+
+ return 1;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ SpeechNormalizerContext *s = ctx->priv;
+ int ret, 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->queue.available > 0) {
+ AVFrame *in = ff_bufqueue_peek(&s->queue, 0);
+ const int nb_samples = available_samples(ctx);
+
+ if (nb_samples >= in->nb_samples || s->eof) {
+ ff_filter_set_ready(ctx, 10);
+ return 0;
+ }
+ }
+
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SpeechNormalizerContext *s = ctx->priv;
+
+ s->max_period = inlink->sample_rate / 10;
+
+ s->prev_gain = 1.;
+ s->cc = av_calloc(inlink->channels, sizeof(*s->cc));
+ if (!s->cc)
+ return AVERROR(ENOMEM);
+
+ for (int ch = 0; ch < inlink->channels; ch++) {
+ ChannelContext *cc = &s->cc[ch];
+
+ cc->state = -1;
+ cc->gain_state = 1.;
+ }
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_FLTP:
+ s->analyze_channel = analyze_channel_flt;
+ s->filter_channels[0] = filter_channels_flt;
+ s->filter_channels[1] = filter_link_channels_flt;
+ break;
+ case AV_SAMPLE_FMT_DBLP:
+ s->analyze_channel = analyze_channel_dbl;
+ s->filter_channels[0] = filter_channels_dbl;
+ s->filter_channels[1] = filter_link_channels_dbl;
+ break;
+ default:
+ av_assert0(0);
+ }
+
+ return 0;
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+ char *res, int res_len, int flags)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+ int link = s->link;
+ int ret;
+
+ ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
+ if (ret < 0)
+ return ret;
+ if (link != s->link)
+ s->prev_gain = 1.;
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ SpeechNormalizerContext *s = ctx->priv;
+
+ ff_bufqueue_discard_all(&s->queue);
+ av_freep(&s->cc);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad 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 = inputs,
+ .outputs = outputs,
+ .process_command = process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 4c671be329..fbfd8989c6 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -131,6 +131,7 @@ 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;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 7112ec8942..23c9d374ad 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 89
+#define LIBAVFILTER_VERSION_MINOR 90
#define LIBAVFILTER_VERSION_MICRO 100
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-10 15:38:45 +0000