diff --git a/Changelog b/Changelog index d8f151b02d..fcdabce2f2 100644 --- a/Changelog +++ b/Changelog @@ -45,6 +45,7 @@ version : - 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 + +#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