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1			/*
2			* Copyright (c) 2020 Paul B Mahol
3			*
4			* Speech Normalizer
5			*
6			* This file is part of FFmpeg.
7			*
8			* FFmpeg is free software; you can redistribute it and/or
9			* modify it under the terms of the GNU Lesser General Public
10			* License as published by the Free Software Foundation; either
11			* version 2.1 of the License, or (at your option) any later version.
12			*
13			* FFmpeg is distributed in the hope that it will be useful,
14			* but WITHOUT ANY WARRANTY; without even the implied warranty of
15			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16			* Lesser General Public License for more details.
17			*
18			* You should have received a copy of the GNU Lesser General Public
19			* License along with FFmpeg; if not, write to the Free Software
20			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21			*/
22
23			/**
24			* @file
25			* Speech Normalizer
26			*/
27
28			#include <float.h>
29
30			#include "libavutil/avassert.h"
31			#include "libavutil/channel_layout.h"
32			#include "libavutil/mem.h"
33			#include "libavutil/opt.h"
34
35			#define FF_BUFQUEUE_SIZE (1024)
36			#include "bufferqueue.h"
37
38			#include "audio.h"
39			#include "avfilter.h"
40			#include "filters.h"
41
42			#define MAX_ITEMS 882000
43			#define MIN_PEAK (1. / 32768.)
44
45			typedef struct PeriodItem {
46			int size;
47			int type;
48			double max_peak;
49			double rms_sum;
50			} PeriodItem;
51
52			typedef struct ChannelContext {
53			int state;
54			int bypass;
55			PeriodItem pi[MAX_ITEMS];
56			double gain_state;
57			double pi_max_peak;
58			double pi_rms_sum;
59			int pi_start;
60			int pi_end;
61			int pi_size;
62			} ChannelContext;
63
64			typedef struct SpeechNormalizerContext {
65			const AVClass *class;
66
67			double rms_value;
68			double peak_value;
69			double max_expansion;
70			double max_compression;
71			double threshold_value;
72			double raise_amount;
73			double fall_amount;
74			char *ch_layout_str;
75			AVChannelLayout ch_layout;
76			int invert;
77			int link;
78
79			ChannelContext *cc;
80			double prev_gain;
81
82			int max_period;
83			int eof;
84			int64_t pts;
85
86			struct FFBufQueue queue;
87
88			void (*analyze_channel)(AVFilterContext *ctx, ChannelContext *cc,
89			const uint8_t *srcp, int nb_samples);
90			void (*filter_channels[2])(AVFilterContext *ctx,
91			AVFrame *in, AVFrame *out, int nb_samples);
92			} SpeechNormalizerContext;
93
94			#define OFFSET(x) offsetof(SpeechNormalizerContext, x)
95			#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
96
97			static const AVOption speechnorm_options[] = {
98			{ "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.95}, 0.0, 1.0, FLAGS },
99			{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.95}, 0.0, 1.0, FLAGS },
100			{ "expansion", "set the max expansion factor", OFFSET(max_expansion), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
101			{ "e", "set the max expansion factor", OFFSET(max_expansion), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
102			{ "compression", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
103			{ "c", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
104			{ "threshold", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
105			{ "t", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
106			{ "raise", "set the expansion raising amount", OFFSET(raise_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
107			{ "r", "set the expansion raising amount", OFFSET(raise_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
108			{ "fall", "set the compression raising amount", OFFSET(fall_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
109			{ "f", "set the compression raising amount", OFFSET(fall_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
110			{ "channels", "set channels to filter", OFFSET(ch_layout_str), AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
111			{ "h", "set channels to filter", OFFSET(ch_layout_str), AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
112			{ "invert", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
113			{ "i", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
114			{ "link", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
115			{ "l", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
116			{ "rms", "set the RMS value", OFFSET(rms_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0.0, 1.0, FLAGS },
117			{ "m", "set the RMS value", OFFSET(rms_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0.0, 1.0, FLAGS },
118			{ NULL }
119			};
120
121			AVFILTER_DEFINE_CLASS(speechnorm);
122
123		✗	static int get_pi_samples(PeriodItem *pi, int start, int end, int remain)
124			{
125			int sum;
126
127		✗	if (pi[start].type == 0)
128		✗	return remain;
129
130		✗	sum = remain;
131		✗	while (start != end) {
132		✗	start++;
133		✗	if (start >= MAX_ITEMS)
134		✗	start = 0;
135		✗	if (pi[start].type == 0)
136		✗	break;
137			av_assert1(pi[start].size > 0);
138		✗	sum += pi[start].size;
139			}
140
141		✗	return sum;
142			}
143
144		✗	static int available_samples(AVFilterContext *ctx)
145			{
146		✗	SpeechNormalizerContext *s = ctx->priv;
147		✗	AVFilterLink *inlink = ctx->inputs[0];
148			int min_pi_nb_samples;
149
150		✗	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);
151		✗	for (int ch = 1; ch < inlink->ch_layout.nb_channels && min_pi_nb_samples > 0; ch++) {
152		✗	ChannelContext *cc = &s->cc[ch];
153
154		✗	min_pi_nb_samples = FFMIN(min_pi_nb_samples, get_pi_samples(cc->pi, cc->pi_start, cc->pi_end, cc->pi_size));
155			}
156
157		✗	return min_pi_nb_samples;
158			}
159
160		✗	static void consume_pi(ChannelContext *cc, int nb_samples)
161			{
162		✗	if (cc->pi_size >= nb_samples) {
163		✗	cc->pi_size -= nb_samples;
164			} else {
165			av_assert1(0);
166			}
167		✗	}
168
169		✗	static double next_gain(AVFilterContext *ctx, double pi_max_peak, int bypass, double state,
170			double pi_rms_sum, int pi_size)
171			{
172		✗	SpeechNormalizerContext *s = ctx->priv;
173		✗	const double compression = 1. / s->max_compression;
174		✗	const int type = s->invert ? pi_max_peak <= s->threshold_value : pi_max_peak >= s->threshold_value;
175		✗	double expansion = FFMIN(s->max_expansion, s->peak_value / pi_max_peak);
176
177		✗	if (s->rms_value > DBL_EPSILON)
178		✗	expansion = FFMIN(expansion, s->rms_value / sqrt(pi_rms_sum / pi_size));
179
180		✗	if (bypass) {
181		✗	return 1.;
182		✗	} else if (type) {
183		✗	return FFMIN(expansion, state + s->raise_amount);
184			} else {
185		✗	return FFMIN(expansion, FFMAX(compression, state - s->fall_amount));
186			}
187			}
188
189		✗	static void next_pi(AVFilterContext *ctx, ChannelContext *cc, int bypass)
190			{
191			av_assert1(cc->pi_size >= 0);
192		✗	if (cc->pi_size == 0) {
193		✗	SpeechNormalizerContext *s = ctx->priv;
194		✗	int start = cc->pi_start;
195
196			av_assert1(cc->pi[start].size > 0);
197		✗	av_assert0(cc->pi[start].type > 0 || s->eof);
198		✗	cc->pi_size = cc->pi[start].size;
199		✗	cc->pi_rms_sum = cc->pi[start].rms_sum;
200		✗	cc->pi_max_peak = cc->pi[start].max_peak;
201			av_assert1(cc->pi_start != cc->pi_end || s->eof);
202		✗	start++;
203		✗	if (start >= MAX_ITEMS)
204		✗	start = 0;
205		✗	cc->pi_start = start;
206		✗	cc->gain_state = next_gain(ctx, cc->pi_max_peak, bypass, cc->gain_state,
207			cc->pi_rms_sum, cc->pi_size);
208			}
209		✗	}
210
211		✗	static double min_gain(AVFilterContext *ctx, ChannelContext *cc, int max_size)
212			{
213		✗	SpeechNormalizerContext *s = ctx->priv;
214		✗	double min_gain = s->max_expansion;
215		✗	double gain_state = cc->gain_state;
216		✗	int size = cc->pi_size;
217		✗	int idx = cc->pi_start;
218
219		✗	min_gain = FFMIN(min_gain, gain_state);
220		✗	while (size <= max_size) {
221		✗	if (idx == cc->pi_end)
222		✗	break;
223		✗	gain_state = next_gain(ctx, cc->pi[idx].max_peak, 0, gain_state,
224			cc->pi[idx].rms_sum, cc->pi[idx].size);
225		✗	min_gain = FFMIN(min_gain, gain_state);
226		✗	size += cc->pi[idx].size;
227		✗	idx++;
228		✗	if (idx >= MAX_ITEMS)
229		✗	idx = 0;
230			}
231
232		✗	return min_gain;
233			}
234
235			#define ANALYZE_CHANNEL(name, ptype, zero, min_peak) \
236			static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
237			const uint8_t *srcp, int nb_samples) \
238			{ \
239			SpeechNormalizerContext *s = ctx->priv; \
240			const ptype *src = (const ptype *)srcp; \
241			const int max_period = s->max_period; \
242			PeriodItem *pi = (PeriodItem *)&cc->pi; \
243			int pi_end = cc->pi_end; \
244			int n = 0; \
245			\
246			if (cc->state < 0) \
247			cc->state = src[0] >= zero; \
248			\
249			while (n < nb_samples) { \
250			ptype new_max_peak; \
251			ptype new_rms_sum; \
252			int new_size; \
253			\
254			if ((cc->state != (src[n] >= zero)) || \
255			(pi[pi_end].size > max_period)) { \
256			ptype max_peak = pi[pi_end].max_peak; \
257			ptype rms_sum = pi[pi_end].rms_sum; \
258			int state = cc->state; \
259			\
260			cc->state = src[n] >= zero; \
261			av_assert1(pi[pi_end].size > 0); \
262			if (max_peak >= min_peak || \
263			pi[pi_end].size > max_period) { \
264			pi[pi_end].type = 1; \
265			pi_end++; \
266			if (pi_end >= MAX_ITEMS) \
267			pi_end = 0; \
268			if (cc->state != state) { \
269			pi[pi_end].max_peak = DBL_MIN; \
270			pi[pi_end].rms_sum = 0.0; \
271			} else { \
272			pi[pi_end].max_peak = max_peak; \
273			pi[pi_end].rms_sum = rms_sum; \
274			} \
275			pi[pi_end].type = 0; \
276			pi[pi_end].size = 0; \
277			av_assert1(pi_end != cc->pi_start); \
278			} \
279			} \
280			\
281			new_max_peak = pi[pi_end].max_peak; \
282			new_rms_sum = pi[pi_end].rms_sum; \
283			new_size = pi[pi_end].size; \
284			if (cc->state) { \
285			while (src[n] >= zero) { \
286			new_max_peak = FFMAX(new_max_peak, src[n]); \
287			new_rms_sum += src[n] * src[n]; \
288			new_size++; \
289			n++; \
290			if (n >= nb_samples) \
291			break; \
292			} \
293			} else { \
294			while (src[n] < zero) { \
295			new_max_peak = FFMAX(new_max_peak, -src[n]); \
296			new_rms_sum += src[n] * src[n]; \
297			new_size++; \
298			n++; \
299			if (n >= nb_samples) \
300			break; \
301			} \
302			} \
303			\
304			pi[pi_end].max_peak = new_max_peak; \
305			pi[pi_end].rms_sum = new_rms_sum; \
306			pi[pi_end].size = new_size; \
307			} \
308			cc->pi_end = pi_end; \
309			}
310
311		✗	ANALYZE_CHANNEL(dbl, double, 0.0, MIN_PEAK)
312		✗	ANALYZE_CHANNEL(flt, float, 0.f, (float)MIN_PEAK)
313
314			#define FILTER_CHANNELS(name, ptype) \
315			static void filter_channels_## name (AVFilterContext *ctx, \
316			AVFrame *in, AVFrame *out, int nb_samples) \
317			{ \
318			SpeechNormalizerContext *s = ctx->priv; \
319			AVFilterLink *inlink = ctx->inputs[0]; \
320			\
321			for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) { \
322			ChannelContext *cc = &s->cc[ch]; \
323			const ptype *src = (const ptype *)in->extended_data[ch]; \
324			ptype *dst = (ptype *)out->extended_data[ch]; \
325			enum AVChannel channel = av_channel_layout_channel_from_index(&inlink->ch_layout, ch); \
326			const int bypass = av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0; \
327			int n = 0; \
328			\
329			while (n < nb_samples) { \
330			ptype gain; \
331			int size; \
332			\
333			next_pi(ctx, cc, bypass); \
334			size = FFMIN(nb_samples - n, cc->pi_size); \
335			av_assert1(size > 0); \
336			gain = cc->gain_state; \
337			consume_pi(cc, size); \
338			for (int i = n; !ctx->is_disabled && i < n + size; i++) \
339			dst[i] = src[i] * gain; \
340			n += size; \
341			} \
342			} \
343			}
344
345		✗	FILTER_CHANNELS(dbl, double)
346		✗	FILTER_CHANNELS(flt, float)
347
348		✗	static double dlerp(double min, double max, double mix)
349			{
350		✗	return min + (max - min) * mix;
351			}
352
353		✗	static float flerp(float min, float max, float mix)
354			{
355		✗	return min + (max - min) * mix;
356			}
357
358			#define FILTER_LINK_CHANNELS(name, ptype, tlerp) \
359			static void filter_link_channels_## name (AVFilterContext *ctx, \
360			AVFrame *in, AVFrame *out, \
361			int nb_samples) \
362			{ \
363			SpeechNormalizerContext *s = ctx->priv; \
364			AVFilterLink *inlink = ctx->inputs[0]; \
365			int n = 0; \
366			\
367			while (n < nb_samples) { \
368			int min_size = nb_samples - n; \
369			ptype gain = s->max_expansion; \
370			\
371			for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) { \
372			ChannelContext *cc = &s->cc[ch]; \
373			\
374			enum AVChannel channel = av_channel_layout_channel_from_index(&inlink->ch_layout, ch); \
375			cc->bypass = av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0; \
376			\
377			next_pi(ctx, cc, cc->bypass); \
378			min_size = FFMIN(min_size, cc->pi_size); \
379			} \
380			\
381			av_assert1(min_size > 0); \
382			for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) { \
383			ChannelContext *cc = &s->cc[ch]; \
384			\
385			if (cc->bypass) \
386			continue; \
387			gain = FFMIN(gain, min_gain(ctx, cc, min_size)); \
388			} \
389			\
390			for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) { \
391			ChannelContext *cc = &s->cc[ch]; \
392			const ptype *src = (const ptype *)in->extended_data[ch]; \
393			ptype *dst = (ptype *)out->extended_data[ch]; \
394			\
395			consume_pi(cc, min_size); \
396			if (cc->bypass) \
397			continue; \
398			\
399			for (int i = n; !ctx->is_disabled && i < n + min_size; i++) { \
400			ptype g = tlerp(s->prev_gain, gain, (i - n) / (ptype)min_size); \
401			dst[i] = src[i] * g; \
402			} \
403			} \
404			\
405			s->prev_gain = gain; \
406			n += min_size; \
407			} \
408			}
409
410		✗	FILTER_LINK_CHANNELS(dbl, double, dlerp)
411		✗	FILTER_LINK_CHANNELS(flt, float, flerp)
412
413		✗	static int filter_frame(AVFilterContext *ctx)
414			{
415		✗	SpeechNormalizerContext *s = ctx->priv;
416		✗	AVFilterLink *outlink = ctx->outputs[0];
417		✗	AVFilterLink *inlink = ctx->inputs[0];
418			int ret;
419
420		✗	while (s->queue.available > 0) {
421			int min_pi_nb_samples;
422			AVFrame *in, *out;
423
424		✗	in = ff_bufqueue_peek(&s->queue, 0);
425		✗	if (!in)
426		✗	break;
427
428		✗	min_pi_nb_samples = available_samples(ctx);
429		✗	if (min_pi_nb_samples < in->nb_samples && !s->eof)
430		✗	break;
431
432		✗	in = ff_bufqueue_get(&s->queue);
433
434		✗	if (av_frame_is_writable(in)) {
435		✗	out = in;
436			} else {
437		✗	out = ff_get_audio_buffer(outlink, in->nb_samples);
438		✗	if (!out) {
439		✗	av_frame_free(&in);
440		✗	return AVERROR(ENOMEM);
441			}
442		✗	av_frame_copy_props(out, in);
443			}
444
445		✗	s->filter_channels[s->link](ctx, in, out, in->nb_samples);
446
447		✗	s->pts = in->pts + av_rescale_q(in->nb_samples, av_make_q(1, outlink->sample_rate),
448			outlink->time_base);
449
450		✗	if (out != in)
451		✗	av_frame_free(&in);
452		✗	return ff_filter_frame(outlink, out);
453			}
454
455		✗	for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) {
456			AVFrame *in;
457
458		✗	ret = ff_inlink_consume_frame(inlink, &in);
459		✗	if (ret < 0)
460		✗	return ret;
461		✗	if (ret == 0)
462		✗	break;
463
464		✗	ff_bufqueue_add(ctx, &s->queue, in);
465
466		✗	for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
467		✗	ChannelContext *cc = &s->cc[ch];
468
469		✗	s->analyze_channel(ctx, cc, in->extended_data[ch], in->nb_samples);
470			}
471			}
472
473		✗	return 1;
474			}
475
476		✗	static int activate(AVFilterContext *ctx)
477			{
478		✗	AVFilterLink *inlink = ctx->inputs[0];
479		✗	AVFilterLink *outlink = ctx->outputs[0];
480		✗	SpeechNormalizerContext *s = ctx->priv;
481			int ret, status;
482			int64_t pts;
483
484		✗	ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
485		✗	if (ret < 0)
486		✗	return ret;
487		✗	if (strcmp(s->ch_layout_str, "all"))
488		✗	av_channel_layout_from_string(&s->ch_layout,
489		✗	s->ch_layout_str);
490
491		✗	FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
492
493		✗	ret = filter_frame(ctx);
494		✗	if (ret <= 0)
495		✗	return ret;
496
497		✗	if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
498		✗	if (status == AVERROR_EOF)
499		✗	s->eof = 1;
500			}
501
502		✗	if (s->eof && ff_inlink_queued_samples(inlink) == 0 &&
503		✗	s->queue.available == 0) {
504		✗	ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
505		✗	return 0;
506			}
507
508		✗	if (s->queue.available > 0) {
509		✗	AVFrame *in = ff_bufqueue_peek(&s->queue, 0);
510		✗	const int nb_samples = available_samples(ctx);
511
512		✗	if (nb_samples >= in->nb_samples || s->eof) {
513		✗	ff_filter_set_ready(ctx, 10);
514		✗	return 0;
515			}
516			}
517
518		✗	FF_FILTER_FORWARD_WANTED(outlink, inlink);
519
520		✗	return FFERROR_NOT_READY;
521			}
522
523		✗	static int config_input(AVFilterLink *inlink)
524			{
525		✗	AVFilterContext *ctx = inlink->dst;
526		✗	SpeechNormalizerContext *s = ctx->priv;
527
528		✗	s->max_period = inlink->sample_rate / 10;
529
530		✗	s->prev_gain = 1.;
531		✗	s->cc = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->cc));
532		✗	if (!s->cc)
533		✗	return AVERROR(ENOMEM);
534
535		✗	for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
536		✗	ChannelContext *cc = &s->cc[ch];
537
538		✗	cc->state = -1;
539		✗	cc->gain_state = s->max_expansion;
540			}
541
542		✗	switch (inlink->format) {
543		✗	case AV_SAMPLE_FMT_FLTP:
544		✗	s->analyze_channel = analyze_channel_flt;
545		✗	s->filter_channels[0] = filter_channels_flt;
546		✗	s->filter_channels[1] = filter_link_channels_flt;
547		✗	break;
548		✗	case AV_SAMPLE_FMT_DBLP:
549		✗	s->analyze_channel = analyze_channel_dbl;
550		✗	s->filter_channels[0] = filter_channels_dbl;
551		✗	s->filter_channels[1] = filter_link_channels_dbl;
552		✗	break;
553		✗	default:
554			av_assert1(0);
555			}
556
557		✗	return 0;
558			}
559
560		✗	static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
561			char *res, int res_len, int flags)
562			{
563		✗	SpeechNormalizerContext *s = ctx->priv;
564		✗	int link = s->link;
565			int ret;
566
567		✗	ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
568		✗	if (ret < 0)
569		✗	return ret;
570		✗	if (link != s->link)
571		✗	s->prev_gain = 1.;
572
573		✗	return 0;
574			}
575
576		✗	static av_cold void uninit(AVFilterContext *ctx)
577			{
578		✗	SpeechNormalizerContext *s = ctx->priv;
579
580		✗	ff_bufqueue_discard_all(&s->queue);
581		✗	av_channel_layout_uninit(&s->ch_layout);
582		✗	av_freep(&s->cc);
583		✗	}
584
585			static const AVFilterPad inputs[] = {
586			{
587			.name = "default",
588			.type = AVMEDIA_TYPE_AUDIO,
589			.config_props = config_input,
590			},
591			};
592
593			const FFFilter ff_af_speechnorm = {
594			.p.name = "speechnorm",
595			.p.description = NULL_IF_CONFIG_SMALL("Speech Normalizer."),
596			.p.priv_class = &speechnorm_class,
597			.p.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
598			.priv_size = sizeof(SpeechNormalizerContext),
599			.activate = activate,
600			.uninit = uninit,
601			FILTER_INPUTS(inputs),
602			FILTER_OUTPUTS(ff_audio_default_filterpad),
603			FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
604			.process_command = process_command,
605			};
606