FFmpeg coverage

GCC Code Coverage Report

Directory:	../../../ffmpeg/		Exec	Total	Coverage
File:	src/libavfilter/af_anlmdn.c	Lines:	0	180	0.0 %
Date:	2021-01-20 23:14:43	Branches:	0	86	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 <float.h>

#include "libavutil/avassert.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"

#include "af_anlmdndsp.h"

#define WEIGHT_LUT_NBITS 20
#define WEIGHT_LUT_SIZE  (1<<WEIGHT_LUT_NBITS)

#define SQR(x) ((x) * (x))

typedef struct AudioNLMeansContext {
    const AVClass *class;

    float a;
    int64_t pd;
    int64_t rd;
    float m;
    int om;

    float pdiff_lut_scale;
    float weight_lut[WEIGHT_LUT_SIZE];

    int K;
    int S;
    int N;
    int H;

    int offset;
    AVFrame *in;
    AVFrame *cache;

    int64_t pts;

    AVAudioFifo *fifo;
    int eof_left;

    AudioNLMDNDSPContext dsp;
} AudioNLMeansContext;

enum OutModes {
    IN_MODE,
    OUT_MODE,
    NOISE_MODE,
    NB_MODES
};

#define OFFSET(x) offsetof(AudioNLMeansContext, x)
#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption anlmdn_options[] = {
    { "s", "set denoising strength", OFFSET(a),  AV_OPT_TYPE_FLOAT,    {.dbl=0.00001},0.00001, 10, AFT },
    { "p", "set patch duration",     OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
    { "r", "set research duration",  OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
    { "o", "set output mode",        OFFSET(om), AV_OPT_TYPE_INT,      {.i64=OUT_MODE},  0, NB_MODES-1, AFT, "mode" },
    {  "i", "input",                 0,          AV_OPT_TYPE_CONST,    {.i64=IN_MODE},   0,  0, AFT, "mode" },
    {  "o", "output",                0,          AV_OPT_TYPE_CONST,    {.i64=OUT_MODE},  0,  0, AFT, "mode" },
    {  "n", "noise",                 0,          AV_OPT_TYPE_CONST,    {.i64=NOISE_MODE},0,  0, AFT, "mode" },
    { "m", "set smooth factor",      OFFSET(m),  AV_OPT_TYPE_FLOAT,    {.dbl=11.},       1, 15, AFT },
    { NULL }
};

AVFILTER_DEFINE_CLASS(anlmdn);

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats = NULL;
    AVFilterChannelLayouts *layouts = NULL;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_FLTP,
        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 float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
{
    float distance = 0.;

    for (int k = -K; k <= K; k++)
        distance += SQR(f1[k] - f2[k]);

    return distance;
}

static void compute_cache_c(float *cache, const float *f,
                            ptrdiff_t S, ptrdiff_t K,
                            ptrdiff_t i, ptrdiff_t jj)
{
    int v = 0;

    for (int j = jj; j < jj + S; j++, v++)
        cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
}

void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
{
    dsp->compute_distance_ssd = compute_distance_ssd_c;
    dsp->compute_cache        = compute_cache_c;

    if (ARCH_X86)
        ff_anlmdn_init_x86(dsp);
}

static int config_filter(AVFilterContext *ctx)
{
    AudioNLMeansContext *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    int newK, newS, newH, newN;
    AVFrame *new_in, *new_cache;

    newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
    newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);

    newH = newK * 2 + 1;
    newN = newH + (newK + newS) * 2;

    av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);

    if (!s->cache || s->cache->nb_samples < newS * 2) {
        new_cache = ff_get_audio_buffer(outlink, newS * 2);
        if (new_cache) {
            av_frame_free(&s->cache);
            s->cache = new_cache;
        } else {
            return AVERROR(ENOMEM);
        }
    }
    if (!s->cache)
        return AVERROR(ENOMEM);

    s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
    for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
        float w = -i / s->pdiff_lut_scale;

        s->weight_lut[i] = expf(w);
    }

    if (!s->in || s->in->nb_samples < newN) {
        new_in = ff_get_audio_buffer(outlink, newN);
        if (new_in) {
            av_frame_free(&s->in);
            s->in = new_in;
        } else {
            return AVERROR(ENOMEM);
        }
    }
    if (!s->in)
        return AVERROR(ENOMEM);

    s->K = newK;
    s->S = newS;
    s->H = newH;
    s->N = newN;

    return 0;
}

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

    s->eof_left = -1;
    s->pts = AV_NOPTS_VALUE;

    ret = config_filter(ctx);
    if (ret < 0)
        return ret;

    s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
    if (!s->fifo)
        return AVERROR(ENOMEM);

    ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
    if (ret < 0)
        return ret;

    ff_anlmdn_init(&s->dsp);

    return 0;
}

static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
    AudioNLMeansContext *s = ctx->priv;
    AVFrame *out = arg;
    const int S = s->S;
    const int K = s->K;
    const int om = s->om;
    const float *f = (const float *)(s->in->extended_data[ch]) + K;
    float *cache = (float *)s->cache->extended_data[ch];
    const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
    float *dst = (float *)out->extended_data[ch] + s->offset;
    const float smooth = s->m;

    for (int i = S; i < s->H + S; i++) {
        float P = 0.f, Q = 0.f;
        int v = 0;

        if (i == S) {
            for (int j = i - S; j <= i + S; j++) {
                if (i == j)
                    continue;
                cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
            }
        } else {
            s->dsp.compute_cache(cache, f, S, K, i, i - S);
            s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
        }

        for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
            const float distance = cache[j];
            unsigned weight_lut_idx;
            float w;

            if (distance < 0.f) {
                cache[j] = 0.f;
                continue;
            }
            w = distance * sw;
            if (w >= smooth)
                continue;
            weight_lut_idx = w * s->pdiff_lut_scale;
            av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
            w = s->weight_lut[weight_lut_idx];
            P += w * f[i - S + j + (j >= S)];
            Q += w;
        }

        P += f[i];
        Q += 1;

        switch (om) {
        case IN_MODE:    dst[i - S] = f[i];           break;
        case OUT_MODE:   dst[i - S] = P / Q;          break;
        case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
        }
    }

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    AudioNLMeansContext *s = ctx->priv;
    AVFrame *out = NULL;
    int available, wanted, ret;

    if (s->pts == AV_NOPTS_VALUE)
        s->pts = in->pts;

    ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
                              in->nb_samples);
    av_frame_free(&in);

    s->offset = 0;
    available = av_audio_fifo_size(s->fifo);
    wanted = (available / s->H) * s->H;

    if (wanted >= s->H && available >= s->N) {
        out = ff_get_audio_buffer(outlink, wanted);
        if (!out)
            return AVERROR(ENOMEM);
    }

    while (available >= s->N) {
        ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
        if (ret < 0)
            break;

        ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);

        av_audio_fifo_drain(s->fifo, s->H);

        s->offset += s->H;
        available -= s->H;
    }

    if (out) {
        out->pts = s->pts;
        out->nb_samples = s->offset;
        if (s->eof_left >= 0) {
            out->nb_samples = FFMIN(s->eof_left, s->offset);
            s->eof_left -= out->nb_samples;
        }
        s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);

        return ff_filter_frame(outlink, out);
    }

    return ret;
}

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

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

    if (ret == AVERROR_EOF && s->eof_left != 0) {
        AVFrame *in;

        if (s->eof_left < 0)
            s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
        if (s->eof_left <= 0)
            return AVERROR_EOF;
        in = ff_get_audio_buffer(outlink, s->H);
        if (!in)
            return AVERROR(ENOMEM);

        return filter_frame(ctx->inputs[0], in);
    }

    return ret;
}

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;

    ret = config_filter(ctx);
    if (ret < 0)
        return ret;

    return 0;
}

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

    av_audio_fifo_free(s->fifo);
    av_frame_free(&s->in);
    av_frame_free(&s->cache);
}

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

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

AVFilter ff_af_anlmdn = {
    .name          = "anlmdn",
    .description   = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
    .query_formats = query_formats,
    .priv_size     = sizeof(AudioNLMeansContext),
    .priv_class    = &anlmdn_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			* 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 <float.h>
22
23			#include "libavutil/avassert.h"
24			#include "libavutil/audio_fifo.h"
25			#include "libavutil/avstring.h"
26			#include "libavutil/opt.h"
27			#include "avfilter.h"
28			#include "audio.h"
29			#include "formats.h"
30
31			#include "af_anlmdndsp.h"
32
33			#define WEIGHT_LUT_NBITS 20
34			#define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
35
36			#define SQR(x) ((x) * (x))
37
38			typedef struct AudioNLMeansContext {
39			const AVClass *class;
40
41			float a;
42			int64_t pd;
43			int64_t rd;
44			float m;
45			int om;
46
47			float pdiff_lut_scale;
48			float weight_lut[WEIGHT_LUT_SIZE];
49
50			int K;
51			int S;
52			int N;
53			int H;
54
55			int offset;
56			AVFrame *in;
57			AVFrame *cache;
58
59			int64_t pts;
60
61			AVAudioFifo *fifo;
62			int eof_left;
63
64			AudioNLMDNDSPContext dsp;
65			} AudioNLMeansContext;
66
67			enum OutModes {
68			IN_MODE,
69			OUT_MODE,
70			NOISE_MODE,
71			NB_MODES
72			};
73
74			#define OFFSET(x) offsetof(AudioNLMeansContext, x)
75			#define AFT AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM\|AV_OPT_FLAG_RUNTIME_PARAM
76
77			static const AVOption anlmdn_options[] = {
78			{ "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
79			{ "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
80			{ "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
81			{ "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
82			{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" },
83			{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" },
84			{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, "mode" },
85			{ "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AFT },
86			{ NULL }
87			};
88
89			AVFILTER_DEFINE_CLASS(anlmdn);
90
91			static int query_formats(AVFilterContext *ctx)
92			{
93			AVFilterFormats *formats = NULL;
94			AVFilterChannelLayouts *layouts = NULL;
95			static const enum AVSampleFormat sample_fmts[] = {
96			AV_SAMPLE_FMT_FLTP,
97			AV_SAMPLE_FMT_NONE
98			};
99			int ret;
100
101			formats = ff_make_format_list(sample_fmts);
102			if (!formats)
103			return AVERROR(ENOMEM);
104			ret = ff_set_common_formats(ctx, formats);
105			if (ret < 0)
106			return ret;
107
108			layouts = ff_all_channel_counts();
109			if (!layouts)
110			return AVERROR(ENOMEM);
111
112			ret = ff_set_common_channel_layouts(ctx, layouts);
113			if (ret < 0)
114			return ret;
115
116			formats = ff_all_samplerates();
117			return ff_set_common_samplerates(ctx, formats);
118			}
119
120			static float compute_distance_ssd_c(const float f1, const float f2, ptrdiff_t K)
121			{
122			float distance = 0.;
123
124			for (int k = -K; k <= K; k++)
125			distance += SQR(f1[k] - f2[k]);
126
127			return distance;
128			}
129
130			static void compute_cache_c(float cache, const float f,
131			ptrdiff_t S, ptrdiff_t K,
132			ptrdiff_t i, ptrdiff_t jj)
133			{
134			int v = 0;
135
136			for (int j = jj; j < jj + S; j++, v++)
137			cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
138			}
139
140			void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
141			{
142			dsp->compute_distance_ssd = compute_distance_ssd_c;
143			dsp->compute_cache = compute_cache_c;
144
145			if (ARCH_X86)
146			ff_anlmdn_init_x86(dsp);
147			}
148
149			static int config_filter(AVFilterContext *ctx)
150			{
151			AudioNLMeansContext *s = ctx->priv;
152			AVFilterLink *outlink = ctx->outputs[0];
153			int newK, newS, newH, newN;
154			AVFrame new_in, new_cache;
155
156			newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
157			newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
158
159			newH = newK * 2 + 1;
160			newN = newH + (newK + newS) * 2;
161
162			av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
163
164			if (!s->cache \|\| s->cache->nb_samples < newS * 2) {
165			new_cache = ff_get_audio_buffer(outlink, newS * 2);
166			if (new_cache) {
167			av_frame_free(&s->cache);
168			s->cache = new_cache;
169			} else {
170			return AVERROR(ENOMEM);
171			}
172			}
173			if (!s->cache)
174			return AVERROR(ENOMEM);
175
176			s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
177			for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
178			float w = -i / s->pdiff_lut_scale;
179
180			s->weight_lut[i] = expf(w);
181			}
182
183			if (!s->in \|\| s->in->nb_samples < newN) {
184			new_in = ff_get_audio_buffer(outlink, newN);
185			if (new_in) {
186			av_frame_free(&s->in);
187			s->in = new_in;
188			} else {
189			return AVERROR(ENOMEM);
190			}
191			}
192			if (!s->in)
193			return AVERROR(ENOMEM);
194
195			s->K = newK;
196			s->S = newS;
197			s->H = newH;
198			s->N = newN;
199
200			return 0;
201			}
202
203			static int config_output(AVFilterLink *outlink)
204			{
205			AVFilterContext *ctx = outlink->src;
206			AudioNLMeansContext *s = ctx->priv;
207			int ret;
208
209			s->eof_left = -1;
210			s->pts = AV_NOPTS_VALUE;
211
212			ret = config_filter(ctx);
213			if (ret < 0)
214			return ret;
215
216			s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
217			if (!s->fifo)
218			return AVERROR(ENOMEM);
219
220			ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
221			if (ret < 0)
222			return ret;
223
224			ff_anlmdn_init(&s->dsp);
225
226			return 0;
227			}
228
229			static int filter_channel(AVFilterContext ctx, void arg, int ch, int nb_jobs)
230			{
231			AudioNLMeansContext *s = ctx->priv;
232			AVFrame *out = arg;
233			const int S = s->S;
234			const int K = s->K;
235			const int om = s->om;
236			const float f = (const float )(s->in->extended_data[ch]) + K;
237			float cache = (float )s->cache->extended_data[ch];
238			const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
239			float dst = (float )out->extended_data[ch] + s->offset;
240			const float smooth = s->m;
241
242			for (int i = S; i < s->H + S; i++) {
243			float P = 0.f, Q = 0.f;
244			int v = 0;
245
246			if (i == S) {
247			for (int j = i - S; j <= i + S; j++) {
248			if (i == j)
249			continue;
250			cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
251			}
252			} else {
253			s->dsp.compute_cache(cache, f, S, K, i, i - S);
254			s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
255			}
256
257			for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
258			const float distance = cache[j];
259			unsigned weight_lut_idx;
260			float w;
261
262			if (distance < 0.f) {
263			cache[j] = 0.f;
264			continue;
265			}
266			w = distance * sw;
267			if (w >= smooth)
268			continue;
269			weight_lut_idx = w * s->pdiff_lut_scale;
270			av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
271			w = s->weight_lut[weight_lut_idx];
272			P += w * f[i - S + j + (j >= S)];
273			Q += w;
274			}
275
276			P += f[i];
277			Q += 1;
278
279			switch (om) {
280			case IN_MODE: dst[i - S] = f[i]; break;
281			case OUT_MODE: dst[i - S] = P / Q; break;
282			case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
283			}
284			}
285
286			return 0;
287			}
288
289			static int filter_frame(AVFilterLink inlink, AVFrame in)
290			{
291			AVFilterContext *ctx = inlink->dst;
292			AVFilterLink *outlink = ctx->outputs[0];
293			AudioNLMeansContext *s = ctx->priv;
294			AVFrame *out = NULL;
295			int available, wanted, ret;
296
297			if (s->pts == AV_NOPTS_VALUE)
298			s->pts = in->pts;
299
300			ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
301			in->nb_samples);
302			av_frame_free(&in);
303
304			s->offset = 0;
305			available = av_audio_fifo_size(s->fifo);
306			wanted = (available / s->H) * s->H;
307
308			if (wanted >= s->H && available >= s->N) {
309			out = ff_get_audio_buffer(outlink, wanted);
310			if (!out)
311			return AVERROR(ENOMEM);
312			}
313
314			while (available >= s->N) {
315			ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
316			if (ret < 0)
317			break;
318
319			ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);
320
321			av_audio_fifo_drain(s->fifo, s->H);
322
323			s->offset += s->H;
324			available -= s->H;
325			}
326
327			if (out) {
328			out->pts = s->pts;
329			out->nb_samples = s->offset;
330			if (s->eof_left >= 0) {
331			out->nb_samples = FFMIN(s->eof_left, s->offset);
332			s->eof_left -= out->nb_samples;
333			}
334			s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
335
336			return ff_filter_frame(outlink, out);
337			}
338
339			return ret;
340			}
341
342			static int request_frame(AVFilterLink *outlink)
343			{
344			AVFilterContext *ctx = outlink->src;
345			AudioNLMeansContext *s = ctx->priv;
346			int ret;
347
348			ret = ff_request_frame(ctx->inputs[0]);
349
350			if (ret == AVERROR_EOF && s->eof_left != 0) {
351			AVFrame *in;
352
353			if (s->eof_left < 0)
354			s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
355			if (s->eof_left <= 0)
356			return AVERROR_EOF;
357			in = ff_get_audio_buffer(outlink, s->H);
358			if (!in)
359			return AVERROR(ENOMEM);
360
361			return filter_frame(ctx->inputs[0], in);
362			}
363
364			return ret;
365			}
366
367			static int process_command(AVFilterContext ctx, const char cmd, const char *args,
368			char *res, int res_len, int flags)
369			{
370			int ret;
371
372			ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
373			if (ret < 0)
374			return ret;
375
376			ret = config_filter(ctx);
377			if (ret < 0)
378			return ret;
379
380			return 0;
381			}
382
383			static av_cold void uninit(AVFilterContext *ctx)
384			{
385			AudioNLMeansContext *s = ctx->priv;
386
387			av_audio_fifo_free(s->fifo);
388			av_frame_free(&s->in);
389			av_frame_free(&s->cache);
390			}
391
392			static const AVFilterPad inputs[] = {
393			{
394			.name = "default",
395			.type = AVMEDIA_TYPE_AUDIO,
396			.filter_frame = filter_frame,
397			},
398			{ NULL }
399			};
400
401			static const AVFilterPad outputs[] = {
402			{
403			.name = "default",
404			.type = AVMEDIA_TYPE_AUDIO,
405			.config_props = config_output,
406			.request_frame = request_frame,
407			},
408			{ NULL }
409			};
410
411			AVFilter ff_af_anlmdn = {
412			.name = "anlmdn",
413			.description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
414			.query_formats = query_formats,
415			.priv_size = sizeof(AudioNLMeansContext),
416			.priv_class = &anlmdn_class,
417			.uninit = uninit,
418			.inputs = inputs,
419			.outputs = outputs,
420			.process_command = process_command,
421			.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL \|
422			AVFILTER_FLAG_SLICE_THREADS,
423			};