FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_anlms.c	Lines:	0	127	0.0 %
Date:	2020-08-14 10:39:37	Branches:	0	77	0.0 %


/*
 * Copyright (c) 2019 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
 */

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

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

enum OutModes {
    IN_MODE,
    DESIRED_MODE,
    OUT_MODE,
    NOISE_MODE,
    NB_OMODES
};

typedef struct AudioNLMSContext {
    const AVClass *class;

    int order;
    float mu;
    float eps;
    float leakage;
    int output_mode;

    int kernel_size;
    AVFrame *offset;
    AVFrame *delay;
    AVFrame *coeffs;
    AVFrame *tmp;

    AVFrame *frame[2];

    AVFloatDSPContext *fdsp;
} AudioNLMSContext;

#define OFFSET(x) offsetof(AudioNLMSContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption anlms_options[] = {
    { "order",   "set the filter order",   OFFSET(order),   AV_OPT_TYPE_INT,   {.i64=256},  1, INT16_MAX, A },
    { "mu",      "set the filter mu",      OFFSET(mu),      AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 2, AT },
    { "eps",     "set the filter eps",     OFFSET(eps),     AV_OPT_TYPE_FLOAT, {.dbl=1},    0, 1, AT },
    { "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0},    0, 1, AT },
    { "out_mode", "set output mode",       OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" },
    {  "i", "input",                 0,          AV_OPT_TYPE_CONST,    {.i64=IN_MODE},      0, 0, AT, "mode" },
    {  "d", "desired",               0,          AV_OPT_TYPE_CONST,    {.i64=DESIRED_MODE}, 0, 0, AT, "mode" },
    {  "o", "output",                0,          AV_OPT_TYPE_CONST,    {.i64=OUT_MODE},     0, 0, AT, "mode" },
    {  "n", "noise",                 0,          AV_OPT_TYPE_CONST,    {.i64=NOISE_MODE},   0, 0, AT, "mode" },
    { NULL }
};

AVFILTER_DEFINE_CLASS(anlms);

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats;
    AVFilterChannelLayouts *layouts;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_FLTP,
        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 float fir_sample(AudioNLMSContext *s, float sample, float *delay,
                        float *coeffs, float *tmp, int *offset)
{
    const int order = s->order;
    float output;

    delay[*offset] = sample;

    memcpy(tmp, coeffs + order - *offset, order * sizeof(float));

    output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);

    if (--(*offset) < 0)
        *offset = order - 1;

    return output;
}

static float process_sample(AudioNLMSContext *s, float input, float desired,
                            float *delay, float *coeffs, float *tmp, int *offsetp)
{
    const int order = s->order;
    const float leakage = s->leakage;
    const float mu = s->mu;
    const float a = 1.f - leakage * mu;
    float sum, output, e, norm, b;
    int offset = *offsetp;

    delay[offset + order] = input;

    output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
    e = desired - output;

    sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);

    norm = s->eps + sum;
    b = mu * e / norm;

    memcpy(tmp, delay + offset, order * sizeof(float));

    s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);

    s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);

    memcpy(coeffs + order, coeffs, order * sizeof(float));

    switch (s->output_mode) {
    case IN_MODE:       output = input;         break;
    case DESIRED_MODE:  output = desired;       break;
    case OUT_MODE: /*output = output;*/         break;
    case NOISE_MODE: output = desired - output; break;
    }
    return output;
}

static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
    AudioNLMSContext *s = ctx->priv;
    AVFrame *out = arg;
    const int start = (out->channels * jobnr) / nb_jobs;
    const int end = (out->channels * (jobnr+1)) / nb_jobs;

    for (int c = start; c < end; c++) {
        const float *input = (const float *)s->frame[0]->extended_data[c];
        const float *desired = (const float *)s->frame[1]->extended_data[c];
        float *delay = (float *)s->delay->extended_data[c];
        float *coeffs = (float *)s->coeffs->extended_data[c];
        float *tmp = (float *)s->tmp->extended_data[c];
        int *offset = (int *)s->offset->extended_data[c];
        float *output = (float *)out->extended_data[c];

        for (int n = 0; n < out->nb_samples; n++)
            output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
    }

    return 0;
}

static int activate(AVFilterContext *ctx)
{
    AudioNLMSContext *s = ctx->priv;
    int i, ret, status;
    int nb_samples;
    int64_t pts;

    FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);

    nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
                       ff_inlink_queued_samples(ctx->inputs[1]));
    for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
        if (s->frame[i])
            continue;

        if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
            ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
            if (ret < 0)
                return ret;
        }
    }

    if (s->frame[0] && s->frame[1]) {
        AVFrame *out;

        out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
        if (!out) {
            av_frame_free(&s->frame[0]);
            av_frame_free(&s->frame[1]);
            return AVERROR(ENOMEM);
        }

        ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
                                                                       ff_filter_get_nb_threads(ctx)));

        out->pts = s->frame[0]->pts;

        av_frame_free(&s->frame[0]);
        av_frame_free(&s->frame[1]);

        ret = ff_filter_frame(ctx->outputs[0], out);
        if (ret < 0)
            return ret;
    }

    if (!nb_samples) {
        for (i = 0; i < 2; i++) {
            if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
                ff_outlink_set_status(ctx->outputs[0], status, pts);
                return 0;
            }
        }
    }

    if (ff_outlink_frame_wanted(ctx->outputs[0])) {
        for (i = 0; i < 2; i++) {
            if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
                continue;
            ff_inlink_request_frame(ctx->inputs[i]);
            return 0;
        }
    }
    return 0;
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AudioNLMSContext *s = ctx->priv;

    s->kernel_size = FFALIGN(s->order, 16);

    if (!s->offset)
        s->offset = ff_get_audio_buffer(outlink, 1);
    if (!s->delay)
        s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
    if (!s->coeffs)
        s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
    if (!s->tmp)
        s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
    if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
        return AVERROR(ENOMEM);

    return 0;
}

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

    s->fdsp = avpriv_float_dsp_alloc(0);
    if (!s->fdsp)
        return AVERROR(ENOMEM);

    return 0;
}

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

    av_freep(&s->fdsp);
    av_frame_free(&s->delay);
    av_frame_free(&s->coeffs);
    av_frame_free(&s->offset);
    av_frame_free(&s->tmp);
}

static const AVFilterPad inputs[] = {
    {
        .name = "input",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    {
        .name = "desired",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

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

AVFilter ff_af_anlms = {
    .name           = "anlms",
    .description    = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
    .priv_size      = sizeof(AudioNLMSContext),
    .priv_class     = &anlms_class,
    .init           = init,
    .uninit         = uninit,
    .activate       = activate,
    .query_formats  = query_formats,
    .inputs         = inputs,
    .outputs        = outputs,
    .flags          = AVFILTER_FLAG_SLICE_THREADS,
    .process_command = ff_filter_process_command,
};


Generated by: GCOVR (Version 4.2)

Line	Branch	Exec	Source
1			/*
2			* Copyright (c) 2019 Paul B Mahol
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/avassert.h"
22			#include "libavutil/channel_layout.h"
23			#include "libavutil/common.h"
24			#include "libavutil/float_dsp.h"
25			#include "libavutil/opt.h"
26
27			#include "audio.h"
28			#include "avfilter.h"
29			#include "formats.h"
30			#include "filters.h"
31			#include "internal.h"
32
33			enum OutModes {
34			IN_MODE,
35			DESIRED_MODE,
36			OUT_MODE,
37			NOISE_MODE,
38			NB_OMODES
39			};
40
41			typedef struct AudioNLMSContext {
42			const AVClass *class;
43
44			int order;
45			float mu;
46			float eps;
47			float leakage;
48			int output_mode;
49
50			int kernel_size;
51			AVFrame *offset;
52			AVFrame *delay;
53			AVFrame *coeffs;
54			AVFrame *tmp;
55
56			AVFrame *frame[2];
57
58			AVFloatDSPContext *fdsp;
59			} AudioNLMSContext;
60
61			#define OFFSET(x) offsetof(AudioNLMSContext, x)
62			#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
63			#define AT AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM\|AV_OPT_FLAG_RUNTIME_PARAM
64
65			static const AVOption anlms_options[] = {
66			{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
67			{ "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 2, AT },
68			{ "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AT },
69			{ "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, AT },
70			{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" },
71			{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" },
72			{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" },
73			{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
74			{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
75			{ NULL }
76			};
77
78			AVFILTER_DEFINE_CLASS(anlms);
79
80			static int query_formats(AVFilterContext *ctx)
81			{
82			AVFilterFormats *formats;
83			AVFilterChannelLayouts *layouts;
84			static const enum AVSampleFormat sample_fmts[] = {
85			AV_SAMPLE_FMT_FLTP,
86			AV_SAMPLE_FMT_NONE
87			};
88			int ret;
89
90			layouts = ff_all_channel_counts();
91			if (!layouts)
92			return AVERROR(ENOMEM);
93			ret = ff_set_common_channel_layouts(ctx, layouts);
94			if (ret < 0)
95			return ret;
96
97			formats = ff_make_format_list(sample_fmts);
98			if (!formats)
99			return AVERROR(ENOMEM);
100			ret = ff_set_common_formats(ctx, formats);
101			if (ret < 0)
102			return ret;
103
104			formats = ff_all_samplerates();
105			if (!formats)
106			return AVERROR(ENOMEM);
107			return ff_set_common_samplerates(ctx, formats);
108			}
109
110			static float fir_sample(AudioNLMSContext s, float sample, float delay,
111			float coeffs, float tmp, int *offset)
112			{
113			const int order = s->order;
114			float output;
115
116			delay[*offset] = sample;
117
118			memcpy(tmp, coeffs + order - offset, order sizeof(float));
119
120			output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
121
122			if (--(*offset) < 0)
123			*offset = order - 1;
124
125			return output;
126			}
127
128			static float process_sample(AudioNLMSContext *s, float input, float desired,
129			float delay, float coeffs, float tmp, int offsetp)
130			{
131			const int order = s->order;
132			const float leakage = s->leakage;
133			const float mu = s->mu;
134			const float a = 1.f - leakage * mu;
135			float sum, output, e, norm, b;
136			int offset = *offsetp;
137
138			delay[offset + order] = input;
139
140			output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
141			e = desired - output;
142
143			sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
144
145			norm = s->eps + sum;
146			b = mu * e / norm;
147
148			memcpy(tmp, delay + offset, order * sizeof(float));
149
150			s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
151
152			s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
153
154			memcpy(coeffs + order, coeffs, order * sizeof(float));
155
156			switch (s->output_mode) {
157			case IN_MODE: output = input; break;
158			case DESIRED_MODE: output = desired; break;
159			case OUT_MODE: /output = output;/ break;
160			case NOISE_MODE: output = desired - output; break;
161			}
162			return output;
163			}
164
165			static int process_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
166			{
167			AudioNLMSContext *s = ctx->priv;
168			AVFrame *out = arg;
169			const int start = (out->channels * jobnr) / nb_jobs;
170			const int end = (out->channels * (jobnr+1)) / nb_jobs;
171
172			for (int c = start; c < end; c++) {
173			const float input = (const float )s->frame[0]->extended_data[c];
174			const float desired = (const float )s->frame[1]->extended_data[c];
175			float delay = (float )s->delay->extended_data[c];
176			float coeffs = (float )s->coeffs->extended_data[c];
177			float tmp = (float )s->tmp->extended_data[c];
178			int offset = (int )s->offset->extended_data[c];
179			float output = (float )out->extended_data[c];
180
181			for (int n = 0; n < out->nb_samples; n++)
182			output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
183			}
184
185			return 0;
186			}
187
188			static int activate(AVFilterContext *ctx)
189			{
190			AudioNLMSContext *s = ctx->priv;
191			int i, ret, status;
192			int nb_samples;
193			int64_t pts;
194
195			FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
196
197			nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
198			ff_inlink_queued_samples(ctx->inputs[1]));
199			for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
200			if (s->frame[i])
201			continue;
202
203			if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
204			ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
205			if (ret < 0)
206			return ret;
207			}
208			}
209
210			if (s->frame[0] && s->frame[1]) {
211			AVFrame *out;
212
213			out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
214			if (!out) {
215			av_frame_free(&s->frame[0]);
216			av_frame_free(&s->frame[1]);
217			return AVERROR(ENOMEM);
218			}
219
220			ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
221			ff_filter_get_nb_threads(ctx)));
222
223			out->pts = s->frame[0]->pts;
224
225			av_frame_free(&s->frame[0]);
226			av_frame_free(&s->frame[1]);
227
228			ret = ff_filter_frame(ctx->outputs[0], out);
229			if (ret < 0)
230			return ret;
231			}
232
233			if (!nb_samples) {
234			for (i = 0; i < 2; i++) {
235			if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
236			ff_outlink_set_status(ctx->outputs[0], status, pts);
237			return 0;
238			}
239			}
240			}
241
242			if (ff_outlink_frame_wanted(ctx->outputs[0])) {
243			for (i = 0; i < 2; i++) {
244			if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
245			continue;
246			ff_inlink_request_frame(ctx->inputs[i]);
247			return 0;
248			}
249			}
250			return 0;
251			}
252
253			static int config_output(AVFilterLink *outlink)
254			{
255			AVFilterContext *ctx = outlink->src;
256			AudioNLMSContext *s = ctx->priv;
257
258			s->kernel_size = FFALIGN(s->order, 16);
259
260			if (!s->offset)
261			s->offset = ff_get_audio_buffer(outlink, 1);
262			if (!s->delay)
263			s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
264			if (!s->coeffs)
265			s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
266			if (!s->tmp)
267			s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
268			if (!s->delay \|\| !s->coeffs \|\| !s->offset \|\| !s->tmp)
269			return AVERROR(ENOMEM);
270
271			return 0;
272			}
273
274			static av_cold int init(AVFilterContext *ctx)
275			{
276			AudioNLMSContext *s = ctx->priv;
277
278			s->fdsp = avpriv_float_dsp_alloc(0);
279			if (!s->fdsp)
280			return AVERROR(ENOMEM);
281
282			return 0;
283			}
284
285			static av_cold void uninit(AVFilterContext *ctx)
286			{
287			AudioNLMSContext *s = ctx->priv;
288
289			av_freep(&s->fdsp);
290			av_frame_free(&s->delay);
291			av_frame_free(&s->coeffs);
292			av_frame_free(&s->offset);
293			av_frame_free(&s->tmp);
294			}
295
296			static const AVFilterPad inputs[] = {
297			{
298			.name = "input",
299			.type = AVMEDIA_TYPE_AUDIO,
300			},
301			{
302			.name = "desired",
303			.type = AVMEDIA_TYPE_AUDIO,
304			},
305			{ NULL }
306			};
307
308			static const AVFilterPad outputs[] = {
309			{
310			.name = "default",
311			.type = AVMEDIA_TYPE_AUDIO,
312			.config_props = config_output,
313			},
314			{ NULL }
315			};
316
317			AVFilter ff_af_anlms = {
318			.name = "anlms",
319			.description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
320			.priv_size = sizeof(AudioNLMSContext),
321			.priv_class = &anlms_class,
322			.init = init,
323			.uninit = uninit,
324			.activate = activate,
325			.query_formats = query_formats,
326			.inputs = inputs,
327			.outputs = outputs,
328			.flags = AVFILTER_FLAG_SLICE_THREADS,
329			.process_command = ff_filter_process_command,
330			};