FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_alimiter.c	Lines:	130	170	76.5 %
Date:	2020-08-11 16:46:18	Branches:	57	104	54.8 %


/*
 * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
 * Copyright (c) 2015 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
 * Lookahead limiter filter
 */

#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/opt.h"

#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"

typedef struct AudioLimiterContext {
    const AVClass *class;

    double limit;
    double attack;
    double release;
    double att;
    double level_in;
    double level_out;
    int auto_release;
    int auto_level;
    double asc;
    int asc_c;
    int asc_pos;
    double asc_coeff;

    double *buffer;
    int buffer_size;
    int pos;
    int *nextpos;
    double *nextdelta;

    double delta;
    int nextiter;
    int nextlen;
    int asc_changed;
} AudioLimiterContext;

#define OFFSET(x) offsetof(AudioLimiterContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM

static const AVOption alimiter_options[] = {
    { "level_in",  "set input level",  OFFSET(level_in),     AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
    { "level_out", "set output level", OFFSET(level_out),    AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
    { "limit",     "set limit",        OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
    { "attack",    "set attack",       OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
    { "release",   "set release",      OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
    { "asc",       "enable asc",       OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
    { "asc_level", "set asc level",    OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
    { "level",     "auto level",       OFFSET(auto_level),   AV_OPT_TYPE_BOOL,   {.i64=1},      0,    1, A|F },
    { NULL }
};

AVFILTER_DEFINE_CLASS(alimiter);

static av_cold int init(AVFilterContext *ctx)
{
    AudioLimiterContext *s = ctx->priv;

    s->attack   /= 1000.;
    s->release  /= 1000.;
    s->att       = 1.;
    s->asc_pos   = -1;
    s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;

    return 0;
}

static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
                         double peak, double limit, double patt, int asc)
{
    double rdelta = (1.0 - patt) / (sample_rate * release);

    if (asc && s->auto_release && s->asc_c > 0) {
        double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;

        if (a_att > patt) {
            double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);

            if (delta < rdelta)
                rdelta = delta;
        }
    }

    return rdelta;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AudioLimiterContext *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    const double *src = (const double *)in->data[0];
    const int channels = inlink->channels;
    const int buffer_size = s->buffer_size;
    double *dst, *buffer = s->buffer;
    const double release = s->release;
    const double limit = s->limit;
    double *nextdelta = s->nextdelta;
    double level = s->auto_level ? 1 / limit : 1;
    const double level_out = s->level_out;
    const double level_in = s->level_in;
    int *nextpos = s->nextpos;
    AVFrame *out;
    double *buf;
    int n, c, i;

    if (av_frame_is_writable(in)) {
        out = in;
    } else {
        out = ff_get_audio_buffer(outlink, in->nb_samples);
        if (!out) {
            av_frame_free(&in);
            return AVERROR(ENOMEM);
        }
        av_frame_copy_props(out, in);
    }
    dst = (double *)out->data[0];

    for (n = 0; n < in->nb_samples; n++) {
        double peak = 0;

        for (c = 0; c < channels; c++) {
            double sample = src[c] * level_in;

            buffer[s->pos + c] = sample;
            peak = FFMAX(peak, fabs(sample));
        }

        if (s->auto_release && peak > limit) {
            s->asc += peak;
            s->asc_c++;
        }

        if (peak > limit) {
            double patt = FFMIN(limit / peak, 1.);
            double rdelta = get_rdelta(s, release, inlink->sample_rate,
                                       peak, limit, patt, 0);
            double delta = (limit / peak - s->att) / buffer_size * channels;
            int found = 0;

            if (delta < s->delta) {
                s->delta = delta;
                nextpos[0] = s->pos;
                nextpos[1] = -1;
                nextdelta[0] = rdelta;
                s->nextlen = 1;
                s->nextiter= 0;
            } else {
                for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
                    int j = i % buffer_size;
                    double ppeak, pdelta;

                    ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
                            fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
                    pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
                    if (pdelta < nextdelta[j]) {
                        nextdelta[j] = pdelta;
                        found = 1;
                        break;
                    }
                }
                if (found) {
                    s->nextlen = i - s->nextiter + 1;
                    nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
                    nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
                    nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
                    s->nextlen++;
                }
            }
        }

        buf = &s->buffer[(s->pos + channels) % buffer_size];
        peak = 0;
        for (c = 0; c < channels; c++) {
            double sample = buf[c];

            peak = FFMAX(peak, fabs(sample));
        }

        if (s->pos == s->asc_pos && !s->asc_changed)
            s->asc_pos = -1;

        if (s->auto_release && s->asc_pos == -1 && peak > limit) {
            s->asc -= peak;
            s->asc_c--;
        }

        s->att += s->delta;

        for (c = 0; c < channels; c++)
            dst[c] = buf[c] * s->att;

        if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
            if (s->auto_release) {
                s->delta = get_rdelta(s, release, inlink->sample_rate,
                                      peak, limit, s->att, 1);
                if (s->nextlen > 1) {
                    int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
                    double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
                                                            fabs(buffer[pnextpos]) :
                                                            fabs(buffer[pnextpos + 1]);
                    double pdelta = (limit / ppeak - s->att) /
                                    (((buffer_size + pnextpos -
                                    ((s->pos + channels) % buffer_size)) %
                                    buffer_size) / channels);
                    if (pdelta < s->delta)
                        s->delta = pdelta;
                }
            } else {
                s->delta = nextdelta[s->nextiter];
                s->att = limit / peak;
            }

            s->nextlen -= 1;
            nextpos[s->nextiter] = -1;
            s->nextiter = (s->nextiter + 1) % buffer_size;
        }

        if (s->att > 1.) {
            s->att = 1.;
            s->delta = 0.;
            s->nextiter = 0;
            s->nextlen = 0;
            nextpos[0] = -1;
        }

        if (s->att <= 0.) {
            s->att = 0.0000000000001;
            s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
        }

        if (s->att != 1. && (1. - s->att) < 0.0000000000001)
            s->att = 1.;

        if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
            s->delta = 0.;

        for (c = 0; c < channels; c++)
            dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;

        s->pos = (s->pos + channels) % buffer_size;
        src += channels;
        dst += channels;
    }

    if (in != out)
        av_frame_free(&in);

    return ff_filter_frame(outlink, out);
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats;
    AVFilterChannelLayouts *layouts;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_DBL,
        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;
    AudioLimiterContext *s = ctx->priv;
    int obuffer_size;

    obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
    if (obuffer_size < inlink->channels)
        return AVERROR(EINVAL);

    s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
    s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
    s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
    if (!s->buffer || !s->nextdelta || !s->nextpos)
        return AVERROR(ENOMEM);

    memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
    s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
    s->buffer_size -= s->buffer_size % inlink->channels;

    if (s->buffer_size <= 0) {
        av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
        return AVERROR(EINVAL);
    }

    return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    AudioLimiterContext *s = ctx->priv;

    av_freep(&s->buffer);
    av_freep(&s->nextdelta);
    av_freep(&s->nextpos);
}

static const AVFilterPad alimiter_inputs[] = {
    {
        .name         = "main",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
        .config_props = config_input,
    },
    { NULL }
};

static const AVFilterPad alimiter_outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

AVFilter ff_af_alimiter = {
    .name           = "alimiter",
    .description    = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
    .priv_size      = sizeof(AudioLimiterContext),
    .priv_class     = &alimiter_class,
    .init           = init,
    .uninit         = uninit,
    .query_formats  = query_formats,
    .inputs         = alimiter_inputs,
    .outputs        = alimiter_outputs,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
3			* Copyright (c) 2015 Paul B Mahol
4			*
5			* This file is part of FFmpeg.
6			*
7			* FFmpeg is free software; you can redistribute it and/or
8			* modify it under the terms of the GNU Lesser General Public
9			* License as published by the Free Software Foundation; either
10			* version 2.1 of the License, or (at your option) any later version.
11			*
12			* FFmpeg is distributed in the hope that it will be useful,
13			* but WITHOUT ANY WARRANTY; without even the implied warranty of
14			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15			* Lesser General Public License for more details.
16			*
17			* You should have received a copy of the GNU Lesser General Public
18			* License along with FFmpeg; if not, write to the Free Software
19			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20			*/
21
22			/**
23			* @file
24			* Lookahead limiter filter
25			*/
26
27			#include "libavutil/avassert.h"
28			#include "libavutil/channel_layout.h"
29			#include "libavutil/common.h"
30			#include "libavutil/opt.h"
31
32			#include "audio.h"
33			#include "avfilter.h"
34			#include "formats.h"
35			#include "internal.h"
36
37			typedef struct AudioLimiterContext {
38			const AVClass *class;
39
40			double limit;
41			double attack;
42			double release;
43			double att;
44			double level_in;
45			double level_out;
46			int auto_release;
47			int auto_level;
48			double asc;
49			int asc_c;
50			int asc_pos;
51			double asc_coeff;
52
53			double *buffer;
54			int buffer_size;
55			int pos;
56			int *nextpos;
57			double *nextdelta;
58
59			double delta;
60			int nextiter;
61			int nextlen;
62			int asc_changed;
63			} AudioLimiterContext;
64
65			#define OFFSET(x) offsetof(AudioLimiterContext, x)
66			#define A AV_OPT_FLAG_AUDIO_PARAM
67			#define F AV_OPT_FLAG_FILTERING_PARAM
68
69			static const AVOption alimiter_options[] = {
70			{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A\|F },
71			{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A\|F },
72			{ "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A\|F },
73			{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A\|F },
74			{ "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A\|F },
75			{ "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A\|F },
76			{ "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A\|F },
77			{ "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A\|F },
78			{ NULL }
79			};
80
81			AVFILTER_DEFINE_CLASS(alimiter);
82
83		1	static av_cold int init(AVFilterContext *ctx)
84			{
85		1	AudioLimiterContext *s = ctx->priv;
86
87		1	s->attack /= 1000.;
88		1	s->release /= 1000.;
89		1	s->att = 1.;
90		1	s->asc_pos = -1;
91		1	s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
92
93		1	return 0;
94			}
95
96		138816	static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
97			double peak, double limit, double patt, int asc)
98			{
99		138816	double rdelta = (1.0 - patt) / (sample_rate * release);
100
101	✗✓✗✗ ✗✗	138816	if (asc && s->auto_release && s->asc_c > 0) {
102			double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
103
104			if (a_att > patt) {
105			double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
106
107			if (delta < rdelta)
108			rdelta = delta;
109			}
110			}
111
112		138816	return rdelta;
113			}
114
115		259	static int filter_frame(AVFilterLink inlink, AVFrame in)
116			{
117		259	AVFilterContext *ctx = inlink->dst;
118		259	AudioLimiterContext *s = ctx->priv;
119		259	AVFilterLink *outlink = ctx->outputs[0];
120		259	const double src = (const double )in->data[0];
121		259	const int channels = inlink->channels;
122		259	const int buffer_size = s->buffer_size;
123		259	double dst, buffer = s->buffer;
124		259	const double release = s->release;
125		259	const double limit = s->limit;
126		259	double *nextdelta = s->nextdelta;
127	✓✗	259	double level = s->auto_level ? 1 / limit : 1;
128		259	const double level_out = s->level_out;
129		259	const double level_in = s->level_in;
130		259	int *nextpos = s->nextpos;
131			AVFrame *out;
132			double *buf;
133			int n, c, i;
134
135	✓✗	259	if (av_frame_is_writable(in)) {
136		259	out = in;
137			} else {
138			out = ff_get_audio_buffer(outlink, in->nb_samples);
139			if (!out) {
140			av_frame_free(&in);
141			return AVERROR(ENOMEM);
142			}
143			av_frame_copy_props(out, in);
144			}
145		259	dst = (double *)out->data[0];
146
147	✓✓	264859	for (n = 0; n < in->nb_samples; n++) {
148		264600	double peak = 0;
149
150	✓✓	793800	for (c = 0; c < channels; c++) {
151		529200	double sample = src[c] * level_in;
152
153		529200	buffer[s->pos + c] = sample;
154	✓✓	529200	peak = FFMAX(peak, fabs(sample));
155			}
156
157	✗✓✗✗	264600	if (s->auto_release && peak > limit) {
158			s->asc += peak;
159			s->asc_c++;
160			}
161
162	✓✓	264600	if (peak > limit) {
163	✗✓	138816	double patt = FFMIN(limit / peak, 1.);
164		138816	double rdelta = get_rdelta(s, release, inlink->sample_rate,
165			peak, limit, patt, 0);
166		138816	double delta = (limit / peak - s->att) / buffer_size * channels;
167		138816	int found = 0;
168
169	✓✓	138816	if (delta < s->delta) {
170		1669	s->delta = delta;
171		1669	nextpos[0] = s->pos;
172		1669	nextpos[1] = -1;
173		1669	nextdelta[0] = rdelta;
174		1669	s->nextlen = 1;
175		1669	s->nextiter= 0;
176			} else {
177	✓✓	564375	for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
178		439864	int j = i % buffer_size;
179			double ppeak, pdelta;
180
181		879728	ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
182	✓✓	439864	fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
183		439864	pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
184	✓✓	439864	if (pdelta < nextdelta[j]) {
185		12636	nextdelta[j] = pdelta;
186		12636	found = 1;
187		12636	break;
188			}
189			}
190	✓✓	137147	if (found) {
191		12636	s->nextlen = i - s->nextiter + 1;
192		12636	nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
193		12636	nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
194		12636	nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
195		12636	s->nextlen++;
196			}
197			}
198			}
199
200		264600	buf = &s->buffer[(s->pos + channels) % buffer_size];
201		264600	peak = 0;
202	✓✓	793800	for (c = 0; c < channels; c++) {
203		529200	double sample = buf[c];
204
205	✓✓	529200	peak = FFMAX(peak, fabs(sample));
206			}
207
208	✗✓✗✗	264600	if (s->pos == s->asc_pos && !s->asc_changed)
209			s->asc_pos = -1;
210
211	✗✓✗✗ ✗✗	264600	if (s->auto_release && s->asc_pos == -1 && peak > limit) {
212			s->asc -= peak;
213			s->asc_c--;
214			}
215
216		264600	s->att += s->delta;
217
218	✓✓	793800	for (c = 0; c < channels; c++)
219		529200	dst[c] = buf[c] * s->att;
220
221	✓✓	264600	if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
222	✗✓	2019	if (s->auto_release) {
223			s->delta = get_rdelta(s, release, inlink->sample_rate,
224			peak, limit, s->att, 1);
225			if (s->nextlen > 1) {
226			int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
227			double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
228			fabs(buffer[pnextpos]) :
229			fabs(buffer[pnextpos + 1]);
230			double pdelta = (limit / ppeak - s->att) /
231			(((buffer_size + pnextpos -
232			((s->pos + channels) % buffer_size)) %
233			buffer_size) / channels);
234			if (pdelta < s->delta)
235			s->delta = pdelta;
236			}
237			} else {
238		2019	s->delta = nextdelta[s->nextiter];
239		2019	s->att = limit / peak;
240			}
241
242		2019	s->nextlen -= 1;
243		2019	nextpos[s->nextiter] = -1;
244		2019	s->nextiter = (s->nextiter + 1) % buffer_size;
245			}
246
247	✓✓	264600	if (s->att > 1.) {
248		11	s->att = 1.;
249		11	s->delta = 0.;
250		11	s->nextiter = 0;
251		11	s->nextlen = 0;
252		11	nextpos[0] = -1;
253			}
254
255	✗✓	264600	if (s->att <= 0.) {
256			s->att = 0.0000000000001;
257			s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
258			}
259
260	✓✓✓✓	264600	if (s->att != 1. && (1. - s->att) < 0.0000000000001)
261		5	s->att = 1.;
262
263	✓✓✗✓	264600	if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
264			s->delta = 0.;
265
266	✓✓	793800	for (c = 0; c < channels; c++)
267		529200	dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
268
269		264600	s->pos = (s->pos + channels) % buffer_size;
270		264600	src += channels;
271		264600	dst += channels;
272			}
273
274	✗✓	259	if (in != out)
275			av_frame_free(&in);
276
277		259	return ff_filter_frame(outlink, out);
278			}
279
280		1	static int query_formats(AVFilterContext *ctx)
281			{
282			AVFilterFormats *formats;
283			AVFilterChannelLayouts *layouts;
284			static const enum AVSampleFormat sample_fmts[] = {
285			AV_SAMPLE_FMT_DBL,
286			AV_SAMPLE_FMT_NONE
287			};
288			int ret;
289
290		1	layouts = ff_all_channel_counts();
291	✗✓	1	if (!layouts)
292			return AVERROR(ENOMEM);
293		1	ret = ff_set_common_channel_layouts(ctx, layouts);
294	✗✓	1	if (ret < 0)
295			return ret;
296
297		1	formats = ff_make_format_list(sample_fmts);
298	✗✓	1	if (!formats)
299			return AVERROR(ENOMEM);
300		1	ret = ff_set_common_formats(ctx, formats);
301	✗✓	1	if (ret < 0)
302			return ret;
303
304		1	formats = ff_all_samplerates();
305	✗✓	1	if (!formats)
306			return AVERROR(ENOMEM);
307		1	return ff_set_common_samplerates(ctx, formats);
308			}
309
310		1	static int config_input(AVFilterLink *inlink)
311			{
312		1	AVFilterContext *ctx = inlink->dst;
313		1	AudioLimiterContext *s = ctx->priv;
314			int obuffer_size;
315
316		1	obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
317	✗✓	1	if (obuffer_size < inlink->channels)
318			return AVERROR(EINVAL);
319
320		1	s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
321		1	s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
322		1	s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
323	✓✗✓✗ ✗✓	1	if (!s->buffer \|\| !s->nextdelta \|\| !s->nextpos)
324			return AVERROR(ENOMEM);
325
326		1	memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
327		1	s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
328		1	s->buffer_size -= s->buffer_size % inlink->channels;
329
330	✗✓	1	if (s->buffer_size <= 0) {
331			av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
332			return AVERROR(EINVAL);
333			}
334
335		1	return 0;
336			}
337
338		1	static av_cold void uninit(AVFilterContext *ctx)
339			{
340		1	AudioLimiterContext *s = ctx->priv;
341
342		1	av_freep(&s->buffer);
343		1	av_freep(&s->nextdelta);
344		1	av_freep(&s->nextpos);
345		1	}
346
347			static const AVFilterPad alimiter_inputs[] = {
348			{
349			.name = "main",
350			.type = AVMEDIA_TYPE_AUDIO,
351			.filter_frame = filter_frame,
352			.config_props = config_input,
353			},
354			{ NULL }
355			};
356
357			static const AVFilterPad alimiter_outputs[] = {
358			{
359			.name = "default",
360			.type = AVMEDIA_TYPE_AUDIO,
361			},
362			{ NULL }
363			};
364
365			AVFilter ff_af_alimiter = {
366			.name = "alimiter",
367			.description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
368			.priv_size = sizeof(AudioLimiterContext),
369			.priv_class = &alimiter_class,
370			.init = init,
371			.uninit = uninit,
372			.query_formats = query_formats,
373			.inputs = alimiter_inputs,
374			.outputs = alimiter_outputs,
375			};