FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_dcshift.c	Lines:	35	58	60.3 %
Date:	2020-08-11 16:46:18	Branches:	14	34	41.2 %


/*
 * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
 * Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
 *
 * 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
 */

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

typedef struct DCShiftContext {
    const AVClass *class;
    double dcshift;
    double limiterthreshold;
    double limitergain;
} DCShiftContext;

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

static const AVOption dcshift_options[] = {
    { "shift",       "set DC shift",     OFFSET(dcshift),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
    { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(dcshift);

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

    s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));

    return 0;
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterChannelLayouts *layouts;
    AVFilterFormats *formats;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_S32P, 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 filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    AVFrame *out;
    DCShiftContext *s = ctx->priv;
    int i, j;
    double dcshift = s->dcshift;

    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);
    }

    if (s->limitergain > 0) {
        for (i = 0; i < inlink->channels; i++) {
            const int32_t *src = (int32_t *)in->extended_data[i];
            int32_t *dst = (int32_t *)out->extended_data[i];

            for (j = 0; j < in->nb_samples; j++) {
                double d;

                d = src[j];

                if (d > s->limiterthreshold && dcshift > 0) {
                    d = (d - s->limiterthreshold) * s->limitergain /
                             (INT32_MAX - s->limiterthreshold) +
                             s->limiterthreshold + dcshift;
                } else if (d < -s->limiterthreshold && dcshift < 0) {
                    d = (d + s->limiterthreshold) * s->limitergain /
                             (INT32_MAX - s->limiterthreshold) -
                             s->limiterthreshold + dcshift;
                } else {
                    d = dcshift * INT32_MAX + d;
                }

                dst[j] = av_clipl_int32(d);
            }
        }
    } else {
        for (i = 0; i < inlink->channels; i++) {
            const int32_t *src = (int32_t *)in->extended_data[i];
            int32_t *dst = (int32_t *)out->extended_data[i];

            for (j = 0; j < in->nb_samples; j++) {
                double d = dcshift * (INT32_MAX + 1.) + src[j];

                dst[j] = av_clipl_int32(d);
            }
        }
    }

    if (out != in)
        av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}
static const AVFilterPad dcshift_inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
    },
    { NULL }
};

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

AVFilter ff_af_dcshift = {
    .name           = "dcshift",
    .description    = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(DCShiftContext),
    .priv_class     = &dcshift_class,
    .init           = init,
    .inputs         = dcshift_inputs,
    .outputs        = dcshift_outputs,
    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
3			* Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
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			#include "libavutil/opt.h"
23			#include "libavutil/samplefmt.h"
24			#include "avfilter.h"
25			#include "audio.h"
26			#include "internal.h"
27
28			typedef struct DCShiftContext {
29			const AVClass *class;
30			double dcshift;
31			double limiterthreshold;
32			double limitergain;
33			} DCShiftContext;
34
35			#define OFFSET(x) offsetof(DCShiftContext, x)
36			#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
37
38			static const AVOption dcshift_options[] = {
39			{ "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
40			{ "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
41			{ NULL }
42			};
43
44			AVFILTER_DEFINE_CLASS(dcshift);
45
46		1	static av_cold int init(AVFilterContext *ctx)
47			{
48		1	DCShiftContext *s = ctx->priv;
49
50		1	s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
51
52		1	return 0;
53			}
54
55		1	static int query_formats(AVFilterContext *ctx)
56			{
57			AVFilterChannelLayouts *layouts;
58			AVFilterFormats *formats;
59			static const enum AVSampleFormat sample_fmts[] = {
60			AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
61			};
62			int ret;
63
64		1	layouts = ff_all_channel_counts();
65	✗✓	1	if (!layouts)
66			return AVERROR(ENOMEM);
67		1	ret = ff_set_common_channel_layouts(ctx, layouts);
68	✗✓	1	if (ret < 0)
69			return ret;
70
71		1	formats = ff_make_format_list(sample_fmts);
72	✗✓	1	if (!formats)
73			return AVERROR(ENOMEM);
74		1	ret = ff_set_common_formats(ctx, formats);
75	✗✓	1	if (ret < 0)
76			return ret;
77
78		1	formats = ff_all_samplerates();
79	✗✓	1	if (!formats)
80			return AVERROR(ENOMEM);
81		1	return ff_set_common_samplerates(ctx, formats);
82			}
83
84		20	static int filter_frame(AVFilterLink inlink, AVFrame in)
85			{
86		20	AVFilterContext *ctx = inlink->dst;
87		20	AVFilterLink *outlink = ctx->outputs[0];
88			AVFrame *out;
89		20	DCShiftContext *s = ctx->priv;
90			int i, j;
91		20	double dcshift = s->dcshift;
92
93	✓✗	20	if (av_frame_is_writable(in)) {
94		20	out = in;
95			} else {
96			out = ff_get_audio_buffer(outlink, in->nb_samples);
97			if (!out) {
98			av_frame_free(&in);
99			return AVERROR(ENOMEM);
100			}
101			av_frame_copy_props(out, in);
102			}
103
104	✓✗	20	if (s->limitergain > 0) {
105	✓✓	60	for (i = 0; i < inlink->channels; i++) {
106		40	const int32_t src = (int32_t )in->extended_data[i];
107		40	int32_t dst = (int32_t )out->extended_data[i];
108
109	✓✓	41000	for (j = 0; j < in->nb_samples; j++) {
110			double d;
111
112		40960	d = src[j];
113
114	✗✓✗✗	40960	if (d > s->limiterthreshold && dcshift > 0) {
115			d = (d - s->limiterthreshold) * s->limitergain /
116			(INT32_MAX - s->limiterthreshold) +
117			s->limiterthreshold + dcshift;
118	✗✓✗✗	40960	} else if (d < -s->limiterthreshold && dcshift < 0) {
119			d = (d + s->limiterthreshold) * s->limitergain /
120			(INT32_MAX - s->limiterthreshold) -
121			s->limiterthreshold + dcshift;
122			} else {
123		40960	d = dcshift * INT32_MAX + d;
124			}
125
126		40960	dst[j] = av_clipl_int32(d);
127			}
128			}
129			} else {
130			for (i = 0; i < inlink->channels; i++) {
131			const int32_t src = (int32_t )in->extended_data[i];
132			int32_t dst = (int32_t )out->extended_data[i];
133
134			for (j = 0; j < in->nb_samples; j++) {
135			double d = dcshift * (INT32_MAX + 1.) + src[j];
136
137			dst[j] = av_clipl_int32(d);
138			}
139			}
140			}
141
142	✗✓	20	if (out != in)
143			av_frame_free(&in);
144		20	return ff_filter_frame(outlink, out);
145			}
146			static const AVFilterPad dcshift_inputs[] = {
147			{
148			.name = "default",
149			.type = AVMEDIA_TYPE_AUDIO,
150			.filter_frame = filter_frame,
151			},
152			{ NULL }
153			};
154
155			static const AVFilterPad dcshift_outputs[] = {
156			{
157			.name = "default",
158			.type = AVMEDIA_TYPE_AUDIO,
159			},
160			{ NULL }
161			};
162
163			AVFilter ff_af_dcshift = {
164			.name = "dcshift",
165			.description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
166			.query_formats = query_formats,
167			.priv_size = sizeof(DCShiftContext),
168			.priv_class = &dcshift_class,
169			.init = init,
170			.inputs = dcshift_inputs,
171			.outputs = dcshift_outputs,
172			.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
173			};