FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_asubboost.c	Lines:	0	102	0.0 %
Date:	2021-04-14 23:45:22	Branches:	0	36	0.0 %


/*
 * 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/channel_layout.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"

typedef struct ASubBoostContext {
    const AVClass *class;

    double dry_gain;
    double wet_gain;
    double feedback;
    double decay;
    double delay;
    double cutoff;
    double slope;

    double a0, a1, a2;
    double b0, b1, b2;

    int *write_pos;
    int buffer_samples;

    AVFrame *w;
    AVFrame *buffer;
} ASubBoostContext;

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats = NULL;
    AVFilterChannelLayouts *layouts = NULL;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_DBLP,
        AV_SAMPLE_FMT_NONE
    };
    int 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;

    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_all_samplerates();
    return ff_set_common_samplerates(ctx, formats);
}

static int get_coeffs(AVFilterContext *ctx)
{
    ASubBoostContext *s = ctx->priv;
    AVFilterLink *inlink = ctx->inputs[0];
    double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
    double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);

    s->a0 =  1 + alpha;
    s->a1 = -2 * cos(w0);
    s->a2 =  1 - alpha;
    s->b0 = (1 - cos(w0)) / 2;
    s->b1 =  1 - cos(w0);
    s->b2 = (1 - cos(w0)) / 2;

    s->a1 /= s->a0;
    s->a2 /= s->a0;
    s->b0 /= s->a0;
    s->b1 /= s->a0;
    s->b2 /= s->a0;

    s->buffer_samples = inlink->sample_rate * s->delay / 1000;

    return 0;
}

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

    s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
    s->w = ff_get_audio_buffer(inlink, 2);
    s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
    if (!s->buffer || !s->w || !s->write_pos)
        return AVERROR(ENOMEM);

    return get_coeffs(ctx);
}

typedef struct ThreadData {
    AVFrame *in, *out;
} ThreadData;

static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    ASubBoostContext *s = ctx->priv;
    ThreadData *td = arg;
    AVFrame *out = td->out;
    AVFrame *in = td->in;
    const double mix = ctx->is_disabled ? 0. : 1.;
    const double wet = ctx->is_disabled ? 1. : s->wet_gain;
    const double dry = ctx->is_disabled ? 1. : s->dry_gain;
    const double feedback = s->feedback, decay = s->decay;
    const double b0 = s->b0;
    const double b1 = s->b1;
    const double b2 = s->b2;
    const double a1 = -s->a1;
    const double a2 = -s->a2;
    const int start = (in->channels * jobnr) / nb_jobs;
    const int end = (in->channels * (jobnr+1)) / nb_jobs;
    const int buffer_samples = s->buffer_samples;

    for (int ch = start; ch < end; ch++) {
        const double *src = (const double *)in->extended_data[ch];
        double *dst = (double *)out->extended_data[ch];
        double *buffer = (double *)s->buffer->extended_data[ch];
        double *w = (double *)s->w->extended_data[ch];
        int write_pos = s->write_pos[ch];

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

            out_sample = src[n] * b0 + w[0];
            w[0] = b1 * src[n] + w[1] + a1 * out_sample;
            w[1] = b2 * src[n] + a2 * out_sample;

            buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
            dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;

            if (++write_pos >= buffer_samples)
                write_pos = 0;
        }

        s->write_pos[ch] = write_pos;
    }

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    ThreadData td;
    AVFrame *out;

    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);
    }

    td.in = in; td.out = out;
    ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
                                                            ff_filter_get_nb_threads(ctx)));

    if (out != in)
        av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}

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

    av_frame_free(&s->buffer);
    av_frame_free(&s->w);
    av_freep(&s->write_pos);
}

static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                           char *res, int res_len, int flags)
{
    int ret;

    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
    if (ret < 0)
        return ret;

    return get_coeffs(ctx);
}

#define OFFSET(x) offsetof(ASubBoostContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption asubboost_options[] = {
    { "dry",      "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
    { "wet",      "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
    { "decay",    "set decay",    OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
    { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9},      0,   1, FLAGS },
    { "cutoff",   "set cutoff",   OFFSET(cutoff),   AV_OPT_TYPE_DOUBLE, {.dbl=100},     50, 900, FLAGS },
    { "slope",    "set slope",    OFFSET(slope),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001,   1, FLAGS },
    { "delay",    "set delay",    OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=20},       1, 100, FLAGS },
    { NULL }
};

AVFILTER_DEFINE_CLASS(asubboost);

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

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

AVFilter ff_af_asubboost = {
    .name           = "asubboost",
    .description    = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(ASubBoostContext),
    .priv_class     = &asubboost_class,
    .uninit         = uninit,
    .inputs         = inputs,
    .outputs        = outputs,
    .process_command = process_command,
    .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
                       AVFILTER_FLAG_SLICE_THREADS,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* This file is part of FFmpeg.
3			*
4			* FFmpeg is free software; you can redistribute it and/or
5			* modify it under the terms of the GNU Lesser General Public
6			* License as published by the Free Software Foundation; either
7			* version 2.1 of the License, or (at your option) any later version.
8			*
9			* FFmpeg is distributed in the hope that it will be useful,
10			* but WITHOUT ANY WARRANTY; without even the implied warranty of
11			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12			* Lesser General Public License for more details.
13			*
14			* You should have received a copy of the GNU Lesser General Public
15			* License along with FFmpeg; if not, write to the Free Software
16			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17			*/
18
19			#include "libavutil/channel_layout.h"
20			#include "libavutil/ffmath.h"
21			#include "libavutil/opt.h"
22			#include "avfilter.h"
23			#include "audio.h"
24			#include "formats.h"
25
26			typedef struct ASubBoostContext {
27			const AVClass *class;
28
29			double dry_gain;
30			double wet_gain;
31			double feedback;
32			double decay;
33			double delay;
34			double cutoff;
35			double slope;
36
37			double a0, a1, a2;
38			double b0, b1, b2;
39
40			int *write_pos;
41			int buffer_samples;
42
43			AVFrame *w;
44			AVFrame *buffer;
45			} ASubBoostContext;
46
47			static int query_formats(AVFilterContext *ctx)
48			{
49			AVFilterFormats *formats = NULL;
50			AVFilterChannelLayouts *layouts = NULL;
51			static const enum AVSampleFormat sample_fmts[] = {
52			AV_SAMPLE_FMT_DBLP,
53			AV_SAMPLE_FMT_NONE
54			};
55			int ret;
56
57			formats = ff_make_format_list(sample_fmts);
58			if (!formats)
59			return AVERROR(ENOMEM);
60			ret = ff_set_common_formats(ctx, formats);
61			if (ret < 0)
62			return ret;
63
64			layouts = ff_all_channel_counts();
65			if (!layouts)
66			return AVERROR(ENOMEM);
67
68			ret = ff_set_common_channel_layouts(ctx, layouts);
69			if (ret < 0)
70			return ret;
71
72			formats = ff_all_samplerates();
73			return ff_set_common_samplerates(ctx, formats);
74			}
75
76			static int get_coeffs(AVFilterContext *ctx)
77			{
78			ASubBoostContext *s = ctx->priv;
79			AVFilterLink *inlink = ctx->inputs[0];
80			double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
81			double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
82
83			s->a0 = 1 + alpha;
84			s->a1 = -2 * cos(w0);
85			s->a2 = 1 - alpha;
86			s->b0 = (1 - cos(w0)) / 2;
87			s->b1 = 1 - cos(w0);
88			s->b2 = (1 - cos(w0)) / 2;
89
90			s->a1 /= s->a0;
91			s->a2 /= s->a0;
92			s->b0 /= s->a0;
93			s->b1 /= s->a0;
94			s->b2 /= s->a0;
95
96			s->buffer_samples = inlink->sample_rate * s->delay / 1000;
97
98			return 0;
99			}
100
101			static int config_input(AVFilterLink *inlink)
102			{
103			AVFilterContext *ctx = inlink->dst;
104			ASubBoostContext *s = ctx->priv;
105
106			s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
107			s->w = ff_get_audio_buffer(inlink, 2);
108			s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
109			if (!s->buffer \|\| !s->w \|\| !s->write_pos)
110			return AVERROR(ENOMEM);
111
112			return get_coeffs(ctx);
113			}
114
115			typedef struct ThreadData {
116			AVFrame in, out;
117			} ThreadData;
118
119			static int filter_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
120			{
121			ASubBoostContext *s = ctx->priv;
122			ThreadData *td = arg;
123			AVFrame *out = td->out;
124			AVFrame *in = td->in;
125			const double mix = ctx->is_disabled ? 0. : 1.;
126			const double wet = ctx->is_disabled ? 1. : s->wet_gain;
127			const double dry = ctx->is_disabled ? 1. : s->dry_gain;
128			const double feedback = s->feedback, decay = s->decay;
129			const double b0 = s->b0;
130			const double b1 = s->b1;
131			const double b2 = s->b2;
132			const double a1 = -s->a1;
133			const double a2 = -s->a2;
134			const int start = (in->channels * jobnr) / nb_jobs;
135			const int end = (in->channels * (jobnr+1)) / nb_jobs;
136			const int buffer_samples = s->buffer_samples;
137
138			for (int ch = start; ch < end; ch++) {
139			const double src = (const double )in->extended_data[ch];
140			double dst = (double )out->extended_data[ch];
141			double buffer = (double )s->buffer->extended_data[ch];
142			double w = (double )s->w->extended_data[ch];
143			int write_pos = s->write_pos[ch];
144
145			for (int n = 0; n < in->nb_samples; n++) {
146			double out_sample;
147
148			out_sample = src[n] * b0 + w[0];
149			w[0] = b1 * src[n] + w[1] + a1 * out_sample;
150			w[1] = b2 * src[n] + a2 * out_sample;
151
152			buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
153			dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;
154
155			if (++write_pos >= buffer_samples)
156			write_pos = 0;
157			}
158
159			s->write_pos[ch] = write_pos;
160			}
161
162			return 0;
163			}
164
165			static int filter_frame(AVFilterLink inlink, AVFrame in)
166			{
167			AVFilterContext *ctx = inlink->dst;
168			AVFilterLink *outlink = ctx->outputs[0];
169			ThreadData td;
170			AVFrame *out;
171
172			if (av_frame_is_writable(in)) {
173			out = in;
174			} else {
175			out = ff_get_audio_buffer(outlink, in->nb_samples);
176			if (!out) {
177			av_frame_free(&in);
178			return AVERROR(ENOMEM);
179			}
180			av_frame_copy_props(out, in);
181			}
182
183			td.in = in; td.out = out;
184			ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
185			ff_filter_get_nb_threads(ctx)));
186
187			if (out != in)
188			av_frame_free(&in);
189			return ff_filter_frame(outlink, out);
190			}
191
192			static av_cold void uninit(AVFilterContext *ctx)
193			{
194			ASubBoostContext *s = ctx->priv;
195
196			av_frame_free(&s->buffer);
197			av_frame_free(&s->w);
198			av_freep(&s->write_pos);
199			}
200
201			static int process_command(AVFilterContext ctx, const char cmd, const char *args,
202			char *res, int res_len, int flags)
203			{
204			int ret;
205
206			ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
207			if (ret < 0)
208			return ret;
209
210			return get_coeffs(ctx);
211			}
212
213			#define OFFSET(x) offsetof(ASubBoostContext, x)
214			#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM\|AV_OPT_FLAG_RUNTIME_PARAM
215
216			static const AVOption asubboost_options[] = {
217			{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
218			{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
219			{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
220			{ "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9}, 0, 1, FLAGS },
221			{ "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
222			{ "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
223			{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
224			{ NULL }
225			};
226
227			AVFILTER_DEFINE_CLASS(asubboost);
228
229			static const AVFilterPad inputs[] = {
230			{
231			.name = "default",
232			.type = AVMEDIA_TYPE_AUDIO,
233			.filter_frame = filter_frame,
234			.config_props = config_input,
235			},
236			{ NULL }
237			};
238
239			static const AVFilterPad outputs[] = {
240			{
241			.name = "default",
242			.type = AVMEDIA_TYPE_AUDIO,
243			},
244			{ NULL }
245			};
246
247			AVFilter ff_af_asubboost = {
248			.name = "asubboost",
249			.description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
250			.query_formats = query_formats,
251			.priv_size = sizeof(ASubBoostContext),
252			.priv_class = &asubboost_class,
253			.uninit = uninit,
254			.inputs = inputs,
255			.outputs = outputs,
256			.process_command = process_command,
257			.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL \|
258			AVFILTER_FLAG_SLICE_THREADS,
259			};