FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_compand.c	Lines:	174	299	58.2 %
Date:	2020-10-23 17:01:47	Branches:	69	138	50.0 %


/*
 * Copyright (c) 1999 Chris Bagwell
 * Copyright (c) 1999 Nick Bailey
 * Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
 * Copyright (c) 2013 Paul B Mahol
 * Copyright (c) 2014 Andrew Kelley
 *
 * 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
 * audio compand filter
 */

#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"

typedef struct ChanParam {
    double attack;
    double decay;
    double volume;
} ChanParam;

typedef struct CompandSegment {
    double x, y;
    double a, b;
} CompandSegment;

typedef struct CompandContext {
    const AVClass *class;
    int nb_segments;
    char *attacks, *decays, *points;
    CompandSegment *segments;
    ChanParam *channels;
    double in_min_lin;
    double out_min_lin;
    double curve_dB;
    double gain_dB;
    double initial_volume;
    double delay;
    AVFrame *delay_frame;
    int delay_samples;
    int delay_count;
    int delay_index;
    int64_t pts;

    int (*compand)(AVFilterContext *ctx, AVFrame *frame);
} CompandContext;

#define OFFSET(x) offsetof(CompandContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption compand_options[] = {
    { "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0" }, 0, 0, A },
    { "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, { .str = "0.8" }, 0, 0, A },
    { "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, { .str = "-70/-70|-60/-20|1/0" }, 0, 0, A },
    { "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.01, 900, A },
    { "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 900, A },
    { "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 0, A },
    { "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(compand);

static av_cold int init(AVFilterContext *ctx)
{
    CompandContext *s = ctx->priv;
    s->pts            = AV_NOPTS_VALUE;
    return 0;
}

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

    av_freep(&s->channels);
    av_freep(&s->segments);
    av_frame_free(&s->delay_frame);
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterChannelLayouts *layouts;
    AVFilterFormats *formats;
    static const enum AVSampleFormat sample_fmts[] = {
        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 void count_items(char *item_str, int *nb_items)
{
    char *p;

    *nb_items = 1;
    for (p = item_str; *p; p++) {
        if (*p == ' ' || *p == '|')
            (*nb_items)++;
    }
}

static void update_volume(ChanParam *cp, double in)
{
    double delta = in - cp->volume;

    if (delta > 0.0)
        cp->volume += delta * cp->attack;
    else
        cp->volume += delta * cp->decay;
}

static double get_volume(CompandContext *s, double in_lin)
{
    CompandSegment *cs;
    double in_log, out_log;
    int i;

    if (in_lin < s->in_min_lin)
        return s->out_min_lin;

    in_log = log(in_lin);

    for (i = 1; i < s->nb_segments; i++)
        if (in_log <= s->segments[i].x)
            break;
    cs = &s->segments[i - 1];
    in_log -= cs->x;
    out_log = cs->y + in_log * (cs->a * in_log + cs->b);

    return exp(out_log);
}

static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
{
    CompandContext *s    = ctx->priv;
    AVFilterLink *inlink = ctx->inputs[0];
    const int channels   = inlink->channels;
    const int nb_samples = frame->nb_samples;
    AVFrame *out_frame;
    int chan, i;
    int err;

    if (av_frame_is_writable(frame)) {
        out_frame = frame;
    } else {
        out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
        if (!out_frame) {
            av_frame_free(&frame);
            return AVERROR(ENOMEM);
        }
        err = av_frame_copy_props(out_frame, frame);
        if (err < 0) {
            av_frame_free(&out_frame);
            av_frame_free(&frame);
            return err;
        }
    }

    for (chan = 0; chan < channels; chan++) {
        const double *src = (double *)frame->extended_data[chan];
        double *dst = (double *)out_frame->extended_data[chan];
        ChanParam *cp = &s->channels[chan];

        for (i = 0; i < nb_samples; i++) {
            update_volume(cp, fabs(src[i]));

            dst[i] = src[i] * get_volume(s, cp->volume);
        }
    }

    if (frame != out_frame)
        av_frame_free(&frame);

    return ff_filter_frame(ctx->outputs[0], out_frame);
}

#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
{
    CompandContext *s    = ctx->priv;
    AVFilterLink *inlink = ctx->inputs[0];
    const int channels = inlink->channels;
    const int nb_samples = frame->nb_samples;
    int chan, i, av_uninit(dindex), oindex, av_uninit(count);
    AVFrame *out_frame   = NULL;
    int err;

    if (s->pts == AV_NOPTS_VALUE) {
        s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;
    }

    av_assert1(channels > 0); /* would corrupt delay_count and delay_index */

    for (chan = 0; chan < channels; chan++) {
        AVFrame *delay_frame = s->delay_frame;
        const double *src    = (double *)frame->extended_data[chan];
        double *dbuf         = (double *)delay_frame->extended_data[chan];
        ChanParam *cp        = &s->channels[chan];
        double *dst;

        count  = s->delay_count;
        dindex = s->delay_index;
        for (i = 0, oindex = 0; i < nb_samples; i++) {
            const double in = src[i];
            update_volume(cp, fabs(in));

            if (count >= s->delay_samples) {
                if (!out_frame) {
                    out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples - i);
                    if (!out_frame) {
                        av_frame_free(&frame);
                        return AVERROR(ENOMEM);
                    }
                    err = av_frame_copy_props(out_frame, frame);
                    if (err < 0) {
                        av_frame_free(&out_frame);
                        av_frame_free(&frame);
                        return err;
                    }
                    out_frame->pts = s->pts;
                    s->pts += av_rescale_q(nb_samples - i,
                        (AVRational){ 1, inlink->sample_rate },
                        inlink->time_base);
                }

                dst = (double *)out_frame->extended_data[chan];
                dst[oindex++] = dbuf[dindex] * get_volume(s, cp->volume);
            } else {
                count++;
            }

            dbuf[dindex] = in;
            dindex = MOD(dindex + 1, s->delay_samples);
        }
    }

    s->delay_count = count;
    s->delay_index = dindex;

    av_frame_free(&frame);

    if (out_frame) {
        err = ff_filter_frame(ctx->outputs[0], out_frame);
        return err;
    }

    return 0;
}

static int compand_drain(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    CompandContext *s    = ctx->priv;
    const int channels   = outlink->channels;
    AVFrame *frame       = NULL;
    int chan, i, dindex;

    /* 2048 is to limit output frame size during drain */
    frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
    if (!frame)
        return AVERROR(ENOMEM);
    frame->pts = s->pts;
    s->pts += av_rescale_q(frame->nb_samples,
            (AVRational){ 1, outlink->sample_rate }, outlink->time_base);

    av_assert0(channels > 0);
    for (chan = 0; chan < channels; chan++) {
        AVFrame *delay_frame = s->delay_frame;
        double *dbuf = (double *)delay_frame->extended_data[chan];
        double *dst = (double *)frame->extended_data[chan];
        ChanParam *cp = &s->channels[chan];

        dindex = s->delay_index;
        for (i = 0; i < frame->nb_samples; i++) {
            dst[i] = dbuf[dindex] * get_volume(s, cp->volume);
            dindex = MOD(dindex + 1, s->delay_samples);
        }
    }
    s->delay_count -= frame->nb_samples;
    s->delay_index = dindex;

    return ff_filter_frame(outlink, frame);
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx  = outlink->src;
    CompandContext *s     = ctx->priv;
    const int sample_rate = outlink->sample_rate;
    double radius         = s->curve_dB * M_LN10 / 20.0;
    char *p, *saveptr     = NULL;
    const int channels    = outlink->channels;
    int nb_attacks, nb_decays, nb_points;
    int new_nb_items, num;
    int i;
    int err;


    count_items(s->attacks, &nb_attacks);
    count_items(s->decays, &nb_decays);
    count_items(s->points, &nb_points);

    if (channels <= 0) {
        av_log(ctx, AV_LOG_ERROR, "Invalid number of channels: %d\n", channels);
        return AVERROR(EINVAL);
    }

    if (nb_attacks > channels || nb_decays > channels) {
        av_log(ctx, AV_LOG_WARNING,
                "Number of attacks/decays bigger than number of channels. Ignoring rest of entries.\n");
        nb_attacks = FFMIN(nb_attacks, channels);
        nb_decays  = FFMIN(nb_decays, channels);
    }

    uninit(ctx);

    s->channels = av_mallocz_array(channels, sizeof(*s->channels));
    s->nb_segments = (nb_points + 4) * 2;
    s->segments = av_mallocz_array(s->nb_segments, sizeof(*s->segments));

    if (!s->channels || !s->segments) {
        uninit(ctx);
        return AVERROR(ENOMEM);
    }

    p = s->attacks;
    for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
        char *tstr = av_strtok(p, " |", &saveptr);
        if (!tstr) {
            uninit(ctx);
            return AVERROR(EINVAL);
        }
        p = NULL;
        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;
        if (s->channels[i].attack < 0) {
            uninit(ctx);
            return AVERROR(EINVAL);
        }
    }
    nb_attacks = new_nb_items;

    p = s->decays;
    for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
        char *tstr = av_strtok(p, " |", &saveptr);
        if (!tstr) {
            uninit(ctx);
            return AVERROR(EINVAL);
        }
        p = NULL;
        new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;
        if (s->channels[i].decay < 0) {
            uninit(ctx);
            return AVERROR(EINVAL);
        }
    }
    nb_decays = new_nb_items;

    if (nb_attacks != nb_decays) {
        av_log(ctx, AV_LOG_ERROR,
                "Number of attacks %d differs from number of decays %d.\n",
                nb_attacks, nb_decays);
        uninit(ctx);
        return AVERROR(EINVAL);
    }

    for (i = nb_decays; i < channels; i++) {
        s->channels[i].attack = s->channels[nb_decays - 1].attack;
        s->channels[i].decay = s->channels[nb_decays - 1].decay;
    }

#define S(x) s->segments[2 * ((x) + 1)]
    p = s->points;
    for (i = 0, new_nb_items = 0; i < nb_points; i++) {
        char *tstr = av_strtok(p, " |", &saveptr);
        p = NULL;
        if (!tstr || sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
            av_log(ctx, AV_LOG_ERROR,
                    "Invalid and/or missing input/output value.\n");
            uninit(ctx);
            return AVERROR(EINVAL);
        }
        if (i && S(i - 1).x > S(i).x) {
            av_log(ctx, AV_LOG_ERROR,
                    "Transfer function input values must be increasing.\n");
            uninit(ctx);
            return AVERROR(EINVAL);
        }
        S(i).y -= S(i).x;
        av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
        new_nb_items++;
    }
    num = new_nb_items;

    /* Add 0,0 if necessary */
    if (num == 0 || S(num - 1).x)
        num++;

#undef S
#define S(x) s->segments[2 * (x)]
    /* Add a tail off segment at the start */
    S(0).x = S(1).x - 2 * s->curve_dB;
    S(0).y = S(1).y;
    num++;

    /* Join adjacent colinear segments */
    for (i = 2; i < num; i++) {
        double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
        double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
        int j;

        if (fabs(g1 - g2))
            continue;
        num--;
        for (j = --i; j < num; j++)
            S(j) = S(j + 1);
    }

    for (i = 0; i < s->nb_segments; i += 2) {
        s->segments[i].y += s->gain_dB;
        s->segments[i].x *= M_LN10 / 20;
        s->segments[i].y *= M_LN10 / 20;
    }

#define L(x) s->segments[i - (x)]
    for (i = 4; i < s->nb_segments; i += 2) {
        double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;

        L(4).a = 0;
        L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);

        L(2).a = 0;
        L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);

        theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
        len = hypot(L(2).x - L(4).x, L(2).y - L(4).y);
        r = FFMIN(radius, len);
        L(3).x = L(2).x - r * cos(theta);
        L(3).y = L(2).y - r * sin(theta);

        theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
        len = hypot(L(0).x - L(2).x, L(0).y - L(2).y);
        r = FFMIN(radius, len / 2);
        x = L(2).x + r * cos(theta);
        y = L(2).y + r * sin(theta);

        cx = (L(3).x + L(2).x + x) / 3;
        cy = (L(3).y + L(2).y + y) / 3;

        L(2).x = x;
        L(2).y = y;

        in1  = cx - L(3).x;
        out1 = cy - L(3).y;
        in2  = L(2).x - L(3).x;
        out2 = L(2).y - L(3).y;
        L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
        L(3).b = out1 / in1 - L(3).a * in1;
    }
    L(3).x = 0;
    L(3).y = L(2).y;

    s->in_min_lin  = exp(s->segments[1].x);
    s->out_min_lin = exp(s->segments[1].y);

    for (i = 0; i < channels; i++) {
        ChanParam *cp = &s->channels[i];

        if (cp->attack > 1.0 / sample_rate)
            cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
        else
            cp->attack = 1.0;
        if (cp->decay > 1.0 / sample_rate)
            cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
        else
            cp->decay = 1.0;
        cp->volume = ff_exp10(s->initial_volume / 20);
    }

    s->delay_samples = s->delay * sample_rate;
    if (s->delay_samples <= 0) {
        s->compand = compand_nodelay;
        return 0;
    }

    s->delay_frame = av_frame_alloc();
    if (!s->delay_frame) {
        uninit(ctx);
        return AVERROR(ENOMEM);
    }

    s->delay_frame->format         = outlink->format;
    s->delay_frame->nb_samples     = s->delay_samples;
    s->delay_frame->channel_layout = outlink->channel_layout;

    err = av_frame_get_buffer(s->delay_frame, 0);
    if (err)
        return err;

    s->compand = compand_delay;
    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
    AVFilterContext *ctx = inlink->dst;
    CompandContext *s    = ctx->priv;

    return s->compand(ctx, frame);
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    CompandContext *s    = ctx->priv;
    int ret = 0;

    ret = ff_request_frame(ctx->inputs[0]);

    if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)
        ret = compand_drain(outlink);

    return ret;
}

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

static const AVFilterPad compand_outputs[] = {
    {
        .name          = "default",
        .request_frame = request_frame,
        .config_props  = config_output,
        .type          = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};


AVFilter ff_af_compand = {
    .name           = "compand",
    .description    = NULL_IF_CONFIG_SMALL(
            "Compress or expand audio dynamic range."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(CompandContext),
    .priv_class     = &compand_class,
    .init           = init,
    .uninit         = uninit,
    .inputs         = compand_inputs,
    .outputs        = compand_outputs,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 1999 Chris Bagwell
3			* Copyright (c) 1999 Nick Bailey
4			* Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
5			* Copyright (c) 2013 Paul B Mahol
6			* Copyright (c) 2014 Andrew Kelley
7			*
8			* This file is part of FFmpeg.
9			*
10			* FFmpeg is free software; you can redistribute it and/or
11			* modify it under the terms of the GNU Lesser General Public
12			* License as published by the Free Software Foundation; either
13			* version 2.1 of the License, or (at your option) any later version.
14			*
15			* FFmpeg is distributed in the hope that it will be useful,
16			* but WITHOUT ANY WARRANTY; without even the implied warranty of
17			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18			* Lesser General Public License for more details.
19			*
20			* You should have received a copy of the GNU Lesser General Public
21			* License along with FFmpeg; if not, write to the Free Software
22			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23			*/
24
25			/**
26			* @file
27			* audio compand filter
28			*/
29
30			#include "libavutil/avassert.h"
31			#include "libavutil/avstring.h"
32			#include "libavutil/ffmath.h"
33			#include "libavutil/opt.h"
34			#include "libavutil/samplefmt.h"
35			#include "audio.h"
36			#include "avfilter.h"
37			#include "internal.h"
38
39			typedef struct ChanParam {
40			double attack;
41			double decay;
42			double volume;
43			} ChanParam;
44
45			typedef struct CompandSegment {
46			double x, y;
47			double a, b;
48			} CompandSegment;
49
50			typedef struct CompandContext {
51			const AVClass *class;
52			int nb_segments;
53			char attacks, decays, *points;
54			CompandSegment *segments;
55			ChanParam *channels;
56			double in_min_lin;
57			double out_min_lin;
58			double curve_dB;
59			double gain_dB;
60			double initial_volume;
61			double delay;
62			AVFrame *delay_frame;
63			int delay_samples;
64			int delay_count;
65			int delay_index;
66			int64_t pts;
67
68			int (compand)(AVFilterContext ctx, AVFrame *frame);
69			} CompandContext;
70
71			#define OFFSET(x) offsetof(CompandContext, x)
72			#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
73
74			static const AVOption compand_options[] = {
75			{ "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0" }, 0, 0, A },
76			{ "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, { .str = "0.8" }, 0, 0, A },
77			{ "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, { .str = "-70/-70\|-60/-20\|1/0" }, 0, 0, A },
78			{ "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.01, 900, A },
79			{ "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 900, A },
80			{ "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 0, A },
81			{ "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20, A },
82			{ NULL }
83			};
84
85			AVFILTER_DEFINE_CLASS(compand);
86
87		2	static av_cold int init(AVFilterContext *ctx)
88			{
89		2	CompandContext *s = ctx->priv;
90		2	s->pts = AV_NOPTS_VALUE;
91		2	return 0;
92			}
93
94		3	static av_cold void uninit(AVFilterContext *ctx)
95			{
96		3	CompandContext *s = ctx->priv;
97
98		3	av_freep(&s->channels);
99		3	av_freep(&s->segments);
100		3	av_frame_free(&s->delay_frame);
101		3	}
102
103		1	static int query_formats(AVFilterContext *ctx)
104			{
105			AVFilterChannelLayouts *layouts;
106			AVFilterFormats *formats;
107			static const enum AVSampleFormat sample_fmts[] = {
108			AV_SAMPLE_FMT_DBLP,
109			AV_SAMPLE_FMT_NONE
110			};
111			int ret;
112
113		1	layouts = ff_all_channel_counts();
114	✗✓	1	if (!layouts)
115			return AVERROR(ENOMEM);
116		1	ret = ff_set_common_channel_layouts(ctx, layouts);
117	✗✓	1	if (ret < 0)
118			return ret;
119
120		1	formats = ff_make_format_list(sample_fmts);
121	✗✓	1	if (!formats)
122			return AVERROR(ENOMEM);
123		1	ret = ff_set_common_formats(ctx, formats);
124	✗✓	1	if (ret < 0)
125			return ret;
126
127		1	formats = ff_all_samplerates();
128	✗✓	1	if (!formats)
129			return AVERROR(ENOMEM);
130		1	return ff_set_common_samplerates(ctx, formats);
131			}
132
133		3	static void count_items(char item_str, int nb_items)
134			{
135			char *p;
136
137		3	*nb_items = 1;
138	✓✓	34	for (p = item_str; *p; p++) {
139	✓✗✓✓	31	if (p == ' ' \|\| p == '\|')
140		3	(*nb_items)++;
141			}
142		3	}
143
144		40960	static void update_volume(ChanParam *cp, double in)
145			{
146		40960	double delta = in - cp->volume;
147
148	✗✓	40960	if (delta > 0.0)
149			cp->volume += delta * cp->attack;
150			else
151		40960	cp->volume += delta * cp->decay;
152		40960	}
153
154		40960	static double get_volume(CompandContext *s, double in_lin)
155			{
156			CompandSegment *cs;
157			double in_log, out_log;
158			int i;
159
160	✗✓	40960	if (in_lin < s->in_min_lin)
161			return s->out_min_lin;
162
163		40960	in_log = log(in_lin);
164
165	✓✗	122966	for (i = 1; i < s->nb_segments; i++)
166	✓✓	122966	if (in_log <= s->segments[i].x)
167		40960	break;
168		40960	cs = &s->segments[i - 1];
169		40960	in_log -= cs->x;
170		40960	out_log = cs->y + in_log * (cs->a * in_log + cs->b);
171
172		40960	return exp(out_log);
173			}
174
175		20	static int compand_nodelay(AVFilterContext ctx, AVFrame frame)
176			{
177		20	CompandContext *s = ctx->priv;
178		20	AVFilterLink *inlink = ctx->inputs[0];
179		20	const int channels = inlink->channels;
180		20	const int nb_samples = frame->nb_samples;
181			AVFrame *out_frame;
182			int chan, i;
183			int err;
184
185	✓✗	20	if (av_frame_is_writable(frame)) {
186		20	out_frame = frame;
187			} else {
188			out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
189			if (!out_frame) {
190			av_frame_free(&frame);
191			return AVERROR(ENOMEM);
192			}
193			err = av_frame_copy_props(out_frame, frame);
194			if (err < 0) {
195			av_frame_free(&out_frame);
196			av_frame_free(&frame);
197			return err;
198			}
199			}
200
201	✓✓	60	for (chan = 0; chan < channels; chan++) {
202		40	const double src = (double )frame->extended_data[chan];
203		40	double dst = (double )out_frame->extended_data[chan];
204		40	ChanParam *cp = &s->channels[chan];
205
206	✓✓	41000	for (i = 0; i < nb_samples; i++) {
207		40960	update_volume(cp, fabs(src[i]));
208
209		40960	dst[i] = src[i] * get_volume(s, cp->volume);
210			}
211			}
212
213	✗✓	20	if (frame != out_frame)
214			av_frame_free(&frame);
215
216		20	return ff_filter_frame(ctx->outputs[0], out_frame);
217			}
218
219			#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
220
221			static int compand_delay(AVFilterContext ctx, AVFrame frame)
222			{
223			CompandContext *s = ctx->priv;
224			AVFilterLink *inlink = ctx->inputs[0];
225			const int channels = inlink->channels;
226			const int nb_samples = frame->nb_samples;
227			int chan, i, av_uninit(dindex), oindex, av_uninit(count);
228			AVFrame *out_frame = NULL;
229			int err;
230
231			if (s->pts == AV_NOPTS_VALUE) {
232			s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;
233			}
234
235			av_assert1(channels > 0); /* would corrupt delay_count and delay_index */
236
237			for (chan = 0; chan < channels; chan++) {
238			AVFrame *delay_frame = s->delay_frame;
239			const double src = (double )frame->extended_data[chan];
240			double dbuf = (double )delay_frame->extended_data[chan];
241			ChanParam *cp = &s->channels[chan];
242			double *dst;
243
244			count = s->delay_count;
245			dindex = s->delay_index;
246			for (i = 0, oindex = 0; i < nb_samples; i++) {
247			const double in = src[i];
248			update_volume(cp, fabs(in));
249
250			if (count >= s->delay_samples) {
251			if (!out_frame) {
252			out_frame = ff_get_audio_buffer(ctx->outputs[0], nb_samples - i);
253			if (!out_frame) {
254			av_frame_free(&frame);
255			return AVERROR(ENOMEM);
256			}
257			err = av_frame_copy_props(out_frame, frame);
258			if (err < 0) {
259			av_frame_free(&out_frame);
260			av_frame_free(&frame);
261			return err;
262			}
263			out_frame->pts = s->pts;
264			s->pts += av_rescale_q(nb_samples - i,
265			(AVRational){ 1, inlink->sample_rate },
266			inlink->time_base);
267			}
268
269			dst = (double *)out_frame->extended_data[chan];
270			dst[oindex++] = dbuf[dindex] * get_volume(s, cp->volume);
271			} else {
272			count++;
273			}
274
275			dbuf[dindex] = in;
276			dindex = MOD(dindex + 1, s->delay_samples);
277			}
278			}
279
280			s->delay_count = count;
281			s->delay_index = dindex;
282
283			av_frame_free(&frame);
284
285			if (out_frame) {
286			err = ff_filter_frame(ctx->outputs[0], out_frame);
287			return err;
288			}
289
290			return 0;
291			}
292
293			static int compand_drain(AVFilterLink *outlink)
294			{
295			AVFilterContext *ctx = outlink->src;
296			CompandContext *s = ctx->priv;
297			const int channels = outlink->channels;
298			AVFrame *frame = NULL;
299			int chan, i, dindex;
300
301			/* 2048 is to limit output frame size during drain */
302			frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
303			if (!frame)
304			return AVERROR(ENOMEM);
305			frame->pts = s->pts;
306			s->pts += av_rescale_q(frame->nb_samples,
307			(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
308
309			av_assert0(channels > 0);
310			for (chan = 0; chan < channels; chan++) {
311			AVFrame *delay_frame = s->delay_frame;
312			double dbuf = (double )delay_frame->extended_data[chan];
313			double dst = (double )frame->extended_data[chan];
314			ChanParam *cp = &s->channels[chan];
315
316			dindex = s->delay_index;
317			for (i = 0; i < frame->nb_samples; i++) {
318			dst[i] = dbuf[dindex] * get_volume(s, cp->volume);
319			dindex = MOD(dindex + 1, s->delay_samples);
320			}
321			}
322			s->delay_count -= frame->nb_samples;
323			s->delay_index = dindex;
324
325			return ff_filter_frame(outlink, frame);
326			}
327
328		1	static int config_output(AVFilterLink *outlink)
329			{
330		1	AVFilterContext *ctx = outlink->src;
331		1	CompandContext *s = ctx->priv;
332		1	const int sample_rate = outlink->sample_rate;
333		1	double radius = s->curve_dB * M_LN10 / 20.0;
334		1	char p, saveptr = NULL;
335		1	const int channels = outlink->channels;
336			int nb_attacks, nb_decays, nb_points;
337			int new_nb_items, num;
338			int i;
339			int err;
340
341
342		1	count_items(s->attacks, &nb_attacks);
343		1	count_items(s->decays, &nb_decays);
344		1	count_items(s->points, &nb_points);
345
346	✗✓	1	if (channels <= 0) {
347			av_log(ctx, AV_LOG_ERROR, "Invalid number of channels: %d\n", channels);
348			return AVERROR(EINVAL);
349			}
350
351	✓✗✗✓	1	if (nb_attacks > channels \|\| nb_decays > channels) {
352			av_log(ctx, AV_LOG_WARNING,
353			"Number of attacks/decays bigger than number of channels. Ignoring rest of entries.\n");
354			nb_attacks = FFMIN(nb_attacks, channels);
355			nb_decays = FFMIN(nb_decays, channels);
356			}
357
358		1	uninit(ctx);
359
360		1	s->channels = av_mallocz_array(channels, sizeof(*s->channels));
361		1	s->nb_segments = (nb_points + 4) * 2;
362		1	s->segments = av_mallocz_array(s->nb_segments, sizeof(*s->segments));
363
364	✓✗✗✓	1	if (!s->channels \|\| !s->segments) {
365			uninit(ctx);
366			return AVERROR(ENOMEM);
367			}
368
369		1	p = s->attacks;
370	✓✓	2	for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
371		1	char *tstr = av_strtok(p, " \|", &saveptr);
372	✗✓	1	if (!tstr) {
373			uninit(ctx);
374			return AVERROR(EINVAL);
375			}
376		1	p = NULL;
377		1	new_nb_items += sscanf(tstr, "%lf", &s->channels[i].attack) == 1;
378	✗✓	1	if (s->channels[i].attack < 0) {
379			uninit(ctx);
380			return AVERROR(EINVAL);
381			}
382			}
383		1	nb_attacks = new_nb_items;
384
385		1	p = s->decays;
386	✓✓	2	for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
387		1	char *tstr = av_strtok(p, " \|", &saveptr);
388	✗✓	1	if (!tstr) {
389			uninit(ctx);
390			return AVERROR(EINVAL);
391			}
392		1	p = NULL;
393		1	new_nb_items += sscanf(tstr, "%lf", &s->channels[i].decay) == 1;
394	✗✓	1	if (s->channels[i].decay < 0) {
395			uninit(ctx);
396			return AVERROR(EINVAL);
397			}
398			}
399		1	nb_decays = new_nb_items;
400
401	✗✓	1	if (nb_attacks != nb_decays) {
402			av_log(ctx, AV_LOG_ERROR,
403			"Number of attacks %d differs from number of decays %d.\n",
404			nb_attacks, nb_decays);
405			uninit(ctx);
406			return AVERROR(EINVAL);
407			}
408
409	✓✓	2	for (i = nb_decays; i < channels; i++) {
410		1	s->channels[i].attack = s->channels[nb_decays - 1].attack;
411		1	s->channels[i].decay = s->channels[nb_decays - 1].decay;
412			}
413
414			#define S(x) s->segments[2 * ((x) + 1)]
415		1	p = s->points;
416	✓✓	5	for (i = 0, new_nb_items = 0; i < nb_points; i++) {
417		4	char *tstr = av_strtok(p, " \|", &saveptr);
418		4	p = NULL;
419	✓✗✗✓	4	if (!tstr \|\| sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
420			av_log(ctx, AV_LOG_ERROR,
421			"Invalid and/or missing input/output value.\n");
422			uninit(ctx);
423			return AVERROR(EINVAL);
424			}
425	✓✓✗✓	4	if (i && S(i - 1).x > S(i).x) {
426			av_log(ctx, AV_LOG_ERROR,
427			"Transfer function input values must be increasing.\n");
428			uninit(ctx);
429			return AVERROR(EINVAL);
430			}
431		4	S(i).y -= S(i).x;
432		4	av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
433		4	new_nb_items++;
434			}
435		1	num = new_nb_items;
436
437			/* Add 0,0 if necessary */
438	✓✗✓✗	1	if (num == 0 \|\| S(num - 1).x)
439		1	num++;
440
441			#undef S
442			#define S(x) s->segments[2 * (x)]
443			/* Add a tail off segment at the start */
444		1	S(0).x = S(1).x - 2 * s->curve_dB;
445		1	S(0).y = S(1).y;
446		1	num++;
447
448			/* Join adjacent colinear segments */
449	✓✓	5	for (i = 2; i < num; i++) {
450		4	double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
451		4	double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
452			int j;
453
454	✓✓	4	if (fabs(g1 - g2))
455		3	continue;
456		1	num--;
457	✓✓	5	for (j = --i; j < num; j++)
458		4	S(j) = S(j + 1);
459			}
460
461	✓✓	9	for (i = 0; i < s->nb_segments; i += 2) {
462		8	s->segments[i].y += s->gain_dB;
463		8	s->segments[i].x *= M_LN10 / 20;
464		8	s->segments[i].y *= M_LN10 / 20;
465			}
466
467			#define L(x) s->segments[i - (x)]
468	✓✓	7	for (i = 4; i < s->nb_segments; i += 2) {
469			double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
470
471		6	L(4).a = 0;
472		6	L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
473
474		6	L(2).a = 0;
475		6	L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
476
477		6	theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
478		6	len = hypot(L(2).x - L(4).x, L(2).y - L(4).y);
479	✓✓	6	r = FFMIN(radius, len);
480		6	L(3).x = L(2).x - r * cos(theta);
481		6	L(3).y = L(2).y - r * sin(theta);
482
483		6	theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
484		6	len = hypot(L(0).x - L(2).x, L(0).y - L(2).y);
485	✓✓	6	r = FFMIN(radius, len / 2);
486		6	x = L(2).x + r * cos(theta);
487		6	y = L(2).y + r * sin(theta);
488
489		6	cx = (L(3).x + L(2).x + x) / 3;
490		6	cy = (L(3).y + L(2).y + y) / 3;
491
492		6	L(2).x = x;
493		6	L(2).y = y;
494
495		6	in1 = cx - L(3).x;
496		6	out1 = cy - L(3).y;
497		6	in2 = L(2).x - L(3).x;
498		6	out2 = L(2).y - L(3).y;
499		6	L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
500		6	L(3).b = out1 / in1 - L(3).a * in1;
501			}
502		1	L(3).x = 0;
503		1	L(3).y = L(2).y;
504
505		1	s->in_min_lin = exp(s->segments[1].x);
506		1	s->out_min_lin = exp(s->segments[1].y);
507
508	✓✓	3	for (i = 0; i < channels; i++) {
509		2	ChanParam *cp = &s->channels[i];
510
511	✗✓	2	if (cp->attack > 1.0 / sample_rate)
512			cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
513			else
514		2	cp->attack = 1.0;
515	✓✗	2	if (cp->decay > 1.0 / sample_rate)
516		2	cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
517			else
518			cp->decay = 1.0;
519		2	cp->volume = ff_exp10(s->initial_volume / 20);
520			}
521
522		1	s->delay_samples = s->delay * sample_rate;
523	✓✗	1	if (s->delay_samples <= 0) {
524		1	s->compand = compand_nodelay;
525		1	return 0;
526			}
527
528			s->delay_frame = av_frame_alloc();
529			if (!s->delay_frame) {
530			uninit(ctx);
531			return AVERROR(ENOMEM);
532			}
533
534			s->delay_frame->format = outlink->format;
535			s->delay_frame->nb_samples = s->delay_samples;
536			s->delay_frame->channel_layout = outlink->channel_layout;
537
538			err = av_frame_get_buffer(s->delay_frame, 0);
539			if (err)
540			return err;
541
542			s->compand = compand_delay;
543			return 0;
544			}
545
546		20	static int filter_frame(AVFilterLink inlink, AVFrame frame)
547			{
548		20	AVFilterContext *ctx = inlink->dst;
549		20	CompandContext *s = ctx->priv;
550
551		20	return s->compand(ctx, frame);
552			}
553
554		20	static int request_frame(AVFilterLink *outlink)
555			{
556		20	AVFilterContext *ctx = outlink->src;
557		20	CompandContext *s = ctx->priv;
558		20	int ret = 0;
559
560		20	ret = ff_request_frame(ctx->inputs[0]);
561
562	✓✓✓✗ ✗✓	20	if (ret == AVERROR_EOF && !ctx->is_disabled && s->delay_count)
563			ret = compand_drain(outlink);
564
565		20	return ret;
566			}
567
568			static const AVFilterPad compand_inputs[] = {
569			{
570			.name = "default",
571			.type = AVMEDIA_TYPE_AUDIO,
572			.filter_frame = filter_frame,
573			},
574			{ NULL }
575			};
576
577			static const AVFilterPad compand_outputs[] = {
578			{
579			.name = "default",
580			.request_frame = request_frame,
581			.config_props = config_output,
582			.type = AVMEDIA_TYPE_AUDIO,
583			},
584			{ NULL }
585			};
586
587
588			AVFilter ff_af_compand = {
589			.name = "compand",
590			.description = NULL_IF_CONFIG_SMALL(
591			"Compress or expand audio dynamic range."),
592			.query_formats = query_formats,
593			.priv_size = sizeof(CompandContext),
594			.priv_class = &compand_class,
595			.init = init,
596			.uninit = uninit,
597			.inputs = compand_inputs,
598			.outputs = compand_outputs,
599			};