FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_flanger.c	Lines:	0	89	0.0 %
Date:	2021-01-20 23:14:43	Branches:	0	28	0.0 %


/*
 * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.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/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
#include "generate_wave_table.h"

#define INTERPOLATION_LINEAR    0
#define INTERPOLATION_QUADRATIC 1

typedef struct FlangerContext {
    const AVClass *class;
    double delay_min;
    double delay_depth;
    double feedback_gain;
    double delay_gain;
    double speed;
    int wave_shape;
    double channel_phase;
    int interpolation;
    double in_gain;
    int max_samples;
    uint8_t **delay_buffer;
    int delay_buf_pos;
    double *delay_last;
    float *lfo;
    int lfo_length;
    int lfo_pos;
} FlangerContext;

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

static const AVOption flanger_options[] = {
    { "delay", "base delay in milliseconds",        OFFSET(delay_min),   AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
    { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
    { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
    { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
    { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
    { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
    { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
    { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
    { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
    { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
    { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
    { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
    { "linear",     NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_LINEAR},    0, 0, A, "itype" },
    { "quadratic",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
    { NULL }
};

AVFILTER_DEFINE_CLASS(flanger);

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

    s->feedback_gain /= 100;
    s->delay_gain    /= 100;
    s->channel_phase /= 100;
    s->delay_min     /= 1000;
    s->delay_depth   /= 1000;
    s->in_gain        = 1 / (1 + s->delay_gain);
    s->delay_gain    /= 1 + s->delay_gain;
    s->delay_gain    *= 1 - fabs(s->feedback_gain);

    return 0;
}

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 int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    FlangerContext *s = ctx->priv;

    s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
    s->lfo_length  = inlink->sample_rate / s->speed;
    s->delay_last  = av_calloc(inlink->channels, sizeof(*s->delay_last));
    s->lfo         = av_calloc(s->lfo_length, sizeof(*s->lfo));
    if (!s->lfo || !s->delay_last)
        return AVERROR(ENOMEM);

    ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
                           rint(s->delay_min * inlink->sample_rate),
                           s->max_samples - 2., 3 * M_PI_2);

    return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
                                              inlink->channels, s->max_samples,
                                              inlink->format, 0);
}

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

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

    for (i = 0; i < frame->nb_samples; i++) {

        s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;

        for (chan = 0; chan < inlink->channels; chan++) {
            double *src = (double *)frame->extended_data[chan];
            double *dst = (double *)out_frame->extended_data[chan];
            double delayed_0, delayed_1;
            double delayed;
            double in, out;
            int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
            double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
            int int_delay = (int)delay;
            double frac_delay = modf(delay, &delay);
            double *delay_buffer = (double *)s->delay_buffer[chan];

            in = src[i];
            delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
                                                           s->feedback_gain;
            delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
            delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];

            if (s->interpolation == INTERPOLATION_LINEAR) {
                delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
            } else {
                double a, b;
                double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
                delayed_2 -= delayed_0;
                delayed_1 -= delayed_0;
                a = delayed_2 * .5 - delayed_1;
                b = delayed_1 *  2 - delayed_2 *.5;
                delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
            }

            s->delay_last[chan] = delayed;
            out = in * s->in_gain + delayed * s->delay_gain;
            dst[i] = out;
        }
        s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
    }

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

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

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

    av_freep(&s->lfo);
    av_freep(&s->delay_last);

    if (s->delay_buffer)
        av_freep(&s->delay_buffer[0]);
    av_freep(&s->delay_buffer);
}

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

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

AVFilter ff_af_flanger = {
    .name          = "flanger",
    .description   = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
    .query_formats = query_formats,
    .priv_size     = sizeof(FlangerContext),
    .priv_class    = &flanger_class,
    .init          = init,
    .uninit        = uninit,
    .inputs        = flanger_inputs,
    .outputs       = flanger_outputs,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
3			*
4			* This file is part of FFmpeg.
5			*
6			* FFmpeg is free software; you can redistribute it and/or
7			* modify it under the terms of the GNU Lesser General Public
8			* License as published by the Free Software Foundation; either
9			* version 2.1 of the License, or (at your option) any later version.
10			*
11			* FFmpeg is distributed in the hope that it will be useful,
12			* but WITHOUT ANY WARRANTY; without even the implied warranty of
13			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14			* Lesser General Public License for more details.
15			*
16			* You should have received a copy of the GNU Lesser General Public
17			* License along with FFmpeg; if not, write to the Free Software
18			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19			*/
20
21			#include "libavutil/avstring.h"
22			#include "libavutil/opt.h"
23			#include "libavutil/samplefmt.h"
24			#include "avfilter.h"
25			#include "audio.h"
26			#include "internal.h"
27			#include "generate_wave_table.h"
28
29			#define INTERPOLATION_LINEAR 0
30			#define INTERPOLATION_QUADRATIC 1
31
32			typedef struct FlangerContext {
33			const AVClass *class;
34			double delay_min;
35			double delay_depth;
36			double feedback_gain;
37			double delay_gain;
38			double speed;
39			int wave_shape;
40			double channel_phase;
41			int interpolation;
42			double in_gain;
43			int max_samples;
44			uint8_t **delay_buffer;
45			int delay_buf_pos;
46			double *delay_last;
47			float *lfo;
48			int lfo_length;
49			int lfo_pos;
50			} FlangerContext;
51
52			#define OFFSET(x) offsetof(FlangerContext, x)
53			#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
54
55			static const AVOption flanger_options[] = {
56			{ "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
57			{ "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
58			{ "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
59			{ "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
60			{ "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
61			{ "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
62			{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
63			{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
64			{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
65			{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
66			{ "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
67			{ "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
68			{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
69			{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
70			{ NULL }
71			};
72
73			AVFILTER_DEFINE_CLASS(flanger);
74
75			static av_cold int init(AVFilterContext *ctx)
76			{
77			FlangerContext *s = ctx->priv;
78
79			s->feedback_gain /= 100;
80			s->delay_gain /= 100;
81			s->channel_phase /= 100;
82			s->delay_min /= 1000;
83			s->delay_depth /= 1000;
84			s->in_gain = 1 / (1 + s->delay_gain);
85			s->delay_gain /= 1 + s->delay_gain;
86			s->delay_gain *= 1 - fabs(s->feedback_gain);
87
88			return 0;
89			}
90
91			static int query_formats(AVFilterContext *ctx)
92			{
93			AVFilterChannelLayouts *layouts;
94			AVFilterFormats *formats;
95			static const enum AVSampleFormat sample_fmts[] = {
96			AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
97			};
98			int ret;
99
100			layouts = ff_all_channel_counts();
101			if (!layouts)
102			return AVERROR(ENOMEM);
103			ret = ff_set_common_channel_layouts(ctx, layouts);
104			if (ret < 0)
105			return ret;
106
107			formats = ff_make_format_list(sample_fmts);
108			if (!formats)
109			return AVERROR(ENOMEM);
110			ret = ff_set_common_formats(ctx, formats);
111			if (ret < 0)
112			return ret;
113
114			formats = ff_all_samplerates();
115			if (!formats)
116			return AVERROR(ENOMEM);
117			return ff_set_common_samplerates(ctx, formats);
118			}
119
120			static int config_input(AVFilterLink *inlink)
121			{
122			AVFilterContext *ctx = inlink->dst;
123			FlangerContext *s = ctx->priv;
124
125			s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
126			s->lfo_length = inlink->sample_rate / s->speed;
127			s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
128			s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
129			if (!s->lfo \|\| !s->delay_last)
130			return AVERROR(ENOMEM);
131
132			ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
133			rint(s->delay_min * inlink->sample_rate),
134			s->max_samples - 2., 3 * M_PI_2);
135
136			return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
137			inlink->channels, s->max_samples,
138			inlink->format, 0);
139			}
140
141			static int filter_frame(AVFilterLink inlink, AVFrame frame)
142			{
143			AVFilterContext *ctx = inlink->dst;
144			FlangerContext *s = ctx->priv;
145			AVFrame *out_frame;
146			int chan, i;
147
148			if (av_frame_is_writable(frame)) {
149			out_frame = frame;
150			} else {
151			out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
152			if (!out_frame) {
153			av_frame_free(&frame);
154			return AVERROR(ENOMEM);
155			}
156			av_frame_copy_props(out_frame, frame);
157			}
158
159			for (i = 0; i < frame->nb_samples; i++) {
160
161			s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
162
163			for (chan = 0; chan < inlink->channels; chan++) {
164			double src = (double )frame->extended_data[chan];
165			double dst = (double )out_frame->extended_data[chan];
166			double delayed_0, delayed_1;
167			double delayed;
168			double in, out;
169			int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
170			double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
171			int int_delay = (int)delay;
172			double frac_delay = modf(delay, &delay);
173			double delay_buffer = (double )s->delay_buffer[chan];
174
175			in = src[i];
176			delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
177			s->feedback_gain;
178			delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
179			delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
180
181			if (s->interpolation == INTERPOLATION_LINEAR) {
182			delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
183			} else {
184			double a, b;
185			double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
186			delayed_2 -= delayed_0;
187			delayed_1 -= delayed_0;
188			a = delayed_2 * .5 - delayed_1;
189			b = delayed_1 * 2 - delayed_2 *.5;
190			delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
191			}
192
193			s->delay_last[chan] = delayed;
194			out = in * s->in_gain + delayed * s->delay_gain;
195			dst[i] = out;
196			}
197			s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
198			}
199
200			if (frame != out_frame)
201			av_frame_free(&frame);
202
203			return ff_filter_frame(ctx->outputs[0], out_frame);
204			}
205
206			static av_cold void uninit(AVFilterContext *ctx)
207			{
208			FlangerContext *s = ctx->priv;
209
210			av_freep(&s->lfo);
211			av_freep(&s->delay_last);
212
213			if (s->delay_buffer)
214			av_freep(&s->delay_buffer[0]);
215			av_freep(&s->delay_buffer);
216			}
217
218			static const AVFilterPad flanger_inputs[] = {
219			{
220			.name = "default",
221			.type = AVMEDIA_TYPE_AUDIO,
222			.config_props = config_input,
223			.filter_frame = filter_frame,
224			},
225			{ NULL }
226			};
227
228			static const AVFilterPad flanger_outputs[] = {
229			{
230			.name = "default",
231			.type = AVMEDIA_TYPE_AUDIO,
232			},
233			{ NULL }
234			};
235
236			AVFilter ff_af_flanger = {
237			.name = "flanger",
238			.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
239			.query_formats = query_formats,
240			.priv_size = sizeof(FlangerContext),
241			.priv_class = &flanger_class,
242			.init = init,
243			.uninit = uninit,
244			.inputs = flanger_inputs,
245			.outputs = flanger_outputs,
246			};