From patchwork Sun Nov 8 23:14:53 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Paul B Mahol X-Patchwork-Id: 23437 Return-Path: X-Original-To: patchwork@ffaux-bg.ffmpeg.org Delivered-To: patchwork@ffaux-bg.ffmpeg.org Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org [79.124.17.100]) by ffaux.localdomain (Postfix) with ESMTP id C4CC144AA40 for ; Mon, 9 Nov 2020 01:15:10 +0200 (EET) Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id A334168BC59; Mon, 9 Nov 2020 01:15:10 +0200 (EET) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail-ej1-f65.google.com (mail-ej1-f65.google.com [209.85.218.65]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id DF86468BA1A for ; Mon, 9 Nov 2020 01:15:04 +0200 (EET) Received: by mail-ej1-f65.google.com with SMTP id s25so9682509ejy.6 for ; Sun, 08 Nov 2020 15:15:04 -0800 (PST) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=from:to:subject:date:message-id; bh=lYGOcCLW5dB4kF9WiJ292aacf6gCYo8BxWI0XDgKYVk=; b=IYfB7AMzDYqLvUY2GaPAm7i1OjUfBb3FATVG6y68Baw5/dTfdOkASyw5yW7CFG8snC gjKyI0rwjPzCdtjAviFi5rUP5SuGUvdkrqBBK2U7Tyma7PixRsWJlk/XbOc4jyl+0ZUM 60dzWVMpJZ/1Aced6n1GFe6EeC4q2RLqOfj0+FzX5XdYpKeURI6lRrUnDwc1lBh1aJm3 YeD44qx1meZo+4y+kFSJgLaI6HB/KO8Ek3s7exIlGerjijVHbczx3QmiwTFRF7IEbeei +n/Y3mBOM7Rw71Rvkl4j+J+Y/W+lVZvaF/pp6rgA4h42OaFmA/ktaM33KdVn1iKRCK+Z JyJg== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:from:to:subject:date:message-id; bh=lYGOcCLW5dB4kF9WiJ292aacf6gCYo8BxWI0XDgKYVk=; b=F/O0gdToyuWva1fJKYyZC8MoQOWqnRJ1CyNtDAib1MjpTFKDBuLuSD97cIAXDlQ+yt OzxxkeLbHIeSwqNtxsdwcSigZ8BIwIU+kl7l0Vvdi/QzuNL4vhsKC84cAFsCfE6Tx+0N AV1W1drp7+JWNeCiYXUaqmqzmKnBQq1Q5f3mn9rHzQayyB/dViSiaDBI0zmmOAG6Bd7Z RJkRsTs8Dl0jXJ9g+7FDveP8vkE+BcUdurt0IlcMyj/00o/KBVF2BHsdeVDMVdLfOSQG K2WFK/p4oSa97ldNhdFMzL5Jmd5visAwEPM7ENzXuWPaQtqRqiWbwc+dBQ3n9XNTA0SR B6Mg== X-Gm-Message-State: AOAM530asrlhgd/k3CIOewo6HUYlmCyGnatiWpsQcOUBrGxR2HFJc15X dXwNXJJ4SUrBrizGN+RPfKXn6vpM3tAX2g== X-Google-Smtp-Source: ABdhPJxJD2mW+tuLucULu0ZP7ZantXEQDkM0+HsiuFMBXrNZzsRfKX7fY0Z8eXw3EjoI8KcDuKpY6A== X-Received: by 2002:a17:907:420d:: with SMTP id oh21mr12715690ejb.429.1604877303676; Sun, 08 Nov 2020 15:15:03 -0800 (PST) Received: from localhost.localdomain ([77.237.107.67]) by smtp.gmail.com with ESMTPSA id ok21sm7293642ejb.96.2020.11.08.15.15.02 for (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Sun, 08 Nov 2020 15:15:03 -0800 (PST) From: Paul B Mahol To: ffmpeg-devel@ffmpeg.org Date: Mon, 9 Nov 2020 00:14:53 +0100 Message-Id: <20201108231453.30308-1-onemda@gmail.com> X-Mailer: git-send-email 2.17.1 Subject: [FFmpeg-devel] [PATCH] avfilter: add speechnorm filter X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Signed-off-by: Paul B Mahol --- doc/filters.texi | 57 +++++ libavfilter/Makefile | 1 + libavfilter/af_speechnorm.c | 441 ++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 500 insertions(+) create mode 100644 libavfilter/af_speechnorm.c diff --git a/doc/filters.texi b/doc/filters.texi index 8380f6cac2..7343632f42 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -5276,6 +5276,63 @@ 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 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 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 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, 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. + +@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 bellow or same as +@option{threshold} option will be expanded otherwise it will be compressed. +@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..75ed60be7b --- /dev/null +++ b/libavfilter/af_speechnorm.c @@ -0,0 +1,441 @@ +/* + * 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 + +#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; + uint64_t channels; + int invert; + + ChannelContext *cc; + + 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_channel)(AVFilterContext *ctx, ChannelContext *cc, + int bypass, uint8_t *srcp, 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 }, + { 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 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 get_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, 1); + 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, 1)); + } + + return min_pi_nb_samples; +} + +static int get_pi_size(AVFilterContext *ctx, ChannelContext *cc, int bypass, int nb_samples) +{ + SpeechNormalizerContext *s = ctx->priv; + int start = cc->pi_start; + const double expansion = FFMIN(s->max_expansion, s->peak_value / cc->pi[start].max_peak); + const double compression = 1. / s->max_compression; + const int type = s->invert ? cc->pi[start].max_peak <= s->threshold_value : cc->pi[start].max_peak > s->threshold_value; + const double gain = type ? expansion : compression; + int size; + + av_assert0(cc->pi[start].size > 0); + size = cc->pi[start].size; + if (size <= nb_samples) { + start++; + if (start >= MAX_ITEMS) + start = 0; + cc->pi_start = start; + } else { + cc->pi[start].size -= nb_samples; + size = nb_samples; + } + + if (bypass) { + cc->gain_state = 1.; + } else if (type) { + cc->gain_state = FFMIN(gain, cc->gain_state + s->raise_amount); + } else { + cc->gain_state = FFMIN(expansion, FFMAX(gain, cc->gain_state - s->fall_amount)); + } + + return size; +} + +#define ANALYZE_CHANNEL(name, ptype, zero, min) \ +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) { \ + cc->state = src[n] >= zero; \ + 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 = 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] >= zero) { \ + while (src[n] >= zero) { \ + 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 >= nb_samples) \ + break; \ + } \ + } else { \ + while (src[n] < zero) { \ + 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 >= nb_samples) \ + break; \ + } \ + } \ + } \ +} + +ANALYZE_CHANNEL(dbl, double, 0.0, DBL_MIN) +ANALYZE_CHANNEL(flt, float, 0.f, FLT_MIN) + +#define FILTER_CHANNEL(name, ptype) \ +static void filter_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \ + int bypass, uint8_t *srcp, int nb_samples) \ +{ \ + ptype *src = (ptype *)srcp; \ + int n = 0; \ + \ + while (n < nb_samples) { \ + const int size = get_pi_size(ctx, cc, bypass, nb_samples - n); \ + const ptype gain = cc->gain_state; \ + \ + for (int i = n; i < n + size; i++) \ + src[i] *= gain; \ + n += size; \ + } \ +} + +FILTER_CHANNEL(dbl, double) +FILTER_CHANNEL(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 = get_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); + + for (int ch = 0; ch < inlink->channels; ch++) { + ChannelContext *cc = &s->cc[ch]; + const int bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); + + s->filter_channel(ctx, cc, bypass, in->extended_data[ch], 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 = get_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 / 20; + + 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_channel = filter_channel_flt; + break; + case AV_SAMPLE_FMT_DBLP: + s->analyze_channel = analyze_channel_dbl; + s->filter_channel = filter_channel_dbl; + break; + } + + 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 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;