GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_anlmdn.c Lines: 0 180 0.0 %
Date: 2021-04-18 21:26:34 Branches: 0 86 0.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) 2019 Paul B Mahol
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include <float.h>
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#include "libavutil/avassert.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#include "af_anlmdndsp.h"
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#define WEIGHT_LUT_NBITS 20
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#define WEIGHT_LUT_SIZE  (1<<WEIGHT_LUT_NBITS)
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#define SQR(x) ((x) * (x))
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typedef struct AudioNLMeansContext {
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    const AVClass *class;
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    float a;
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    int64_t pd;
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    int64_t rd;
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    float m;
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    int om;
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    float pdiff_lut_scale;
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    float weight_lut[WEIGHT_LUT_SIZE];
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    int K;
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    int S;
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    int N;
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    int H;
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    int offset;
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    AVFrame *in;
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    AVFrame *cache;
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    int64_t pts;
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    AVAudioFifo *fifo;
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    int eof_left;
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    AudioNLMDNDSPContext dsp;
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} AudioNLMeansContext;
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enum OutModes {
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    IN_MODE,
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    OUT_MODE,
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    NOISE_MODE,
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    NB_MODES
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};
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#define OFFSET(x) offsetof(AudioNLMeansContext, x)
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#define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption anlmdn_options[] = {
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    { "s", "set denoising strength", OFFSET(a),  AV_OPT_TYPE_FLOAT,    {.dbl=0.00001},0.00001, 10, AFT },
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    { "p", "set patch duration",     OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
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    { "r", "set research duration",  OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
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    { "o", "set output mode",        OFFSET(om), AV_OPT_TYPE_INT,      {.i64=OUT_MODE},  0, NB_MODES-1, AFT, "mode" },
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    {  "i", "input",                 0,          AV_OPT_TYPE_CONST,    {.i64=IN_MODE},   0,  0, AFT, "mode" },
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    {  "o", "output",                0,          AV_OPT_TYPE_CONST,    {.i64=OUT_MODE},  0,  0, AFT, "mode" },
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    {  "n", "noise",                 0,          AV_OPT_TYPE_CONST,    {.i64=NOISE_MODE},0,  0, AFT, "mode" },
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    { "m", "set smooth factor",      OFFSET(m),  AV_OPT_TYPE_FLOAT,    {.dbl=11.},       1, 15, AFT },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(anlmdn);
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static int query_formats(AVFilterContext *ctx)
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{
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    AVFilterFormats *formats = NULL;
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    AVFilterChannelLayouts *layouts = NULL;
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    static const enum AVSampleFormat sample_fmts[] = {
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        AV_SAMPLE_FMT_FLTP,
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        AV_SAMPLE_FMT_NONE
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    };
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    int ret;
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    formats = ff_make_format_list(sample_fmts);
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    if (!formats)
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        return AVERROR(ENOMEM);
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    ret = ff_set_common_formats(ctx, formats);
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    if (ret < 0)
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        return ret;
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    layouts = ff_all_channel_counts();
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    if (!layouts)
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        return AVERROR(ENOMEM);
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    ret = ff_set_common_channel_layouts(ctx, layouts);
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    if (ret < 0)
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        return ret;
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    formats = ff_all_samplerates();
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    return ff_set_common_samplerates(ctx, formats);
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}
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static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
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{
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    float distance = 0.;
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    for (int k = -K; k <= K; k++)
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        distance += SQR(f1[k] - f2[k]);
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    return distance;
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}
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static void compute_cache_c(float *cache, const float *f,
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                            ptrdiff_t S, ptrdiff_t K,
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                            ptrdiff_t i, ptrdiff_t jj)
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{
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    int v = 0;
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    for (int j = jj; j < jj + S; j++, v++)
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        cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
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}
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void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
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{
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    dsp->compute_distance_ssd = compute_distance_ssd_c;
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    dsp->compute_cache        = compute_cache_c;
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    if (ARCH_X86)
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        ff_anlmdn_init_x86(dsp);
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}
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static int config_filter(AVFilterContext *ctx)
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{
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    AudioNLMeansContext *s = ctx->priv;
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    AVFilterLink *outlink = ctx->outputs[0];
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    int newK, newS, newH, newN;
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    AVFrame *new_in, *new_cache;
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    newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
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    newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
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    newH = newK * 2 + 1;
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    newN = newH + (newK + newS) * 2;
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    av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
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    if (!s->cache || s->cache->nb_samples < newS * 2) {
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        new_cache = ff_get_audio_buffer(outlink, newS * 2);
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        if (new_cache) {
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            av_frame_free(&s->cache);
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            s->cache = new_cache;
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        } else {
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            return AVERROR(ENOMEM);
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        }
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    }
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    if (!s->cache)
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        return AVERROR(ENOMEM);
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    s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
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    for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
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        float w = -i / s->pdiff_lut_scale;
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        s->weight_lut[i] = expf(w);
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    }
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    if (!s->in || s->in->nb_samples < newN) {
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        new_in = ff_get_audio_buffer(outlink, newN);
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        if (new_in) {
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            av_frame_free(&s->in);
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            s->in = new_in;
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        } else {
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            return AVERROR(ENOMEM);
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        }
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    }
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    if (!s->in)
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        return AVERROR(ENOMEM);
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    s->K = newK;
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    s->S = newS;
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    s->H = newH;
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    s->N = newN;
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    return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    AudioNLMeansContext *s = ctx->priv;
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    int ret;
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    s->eof_left = -1;
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    s->pts = AV_NOPTS_VALUE;
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    ret = config_filter(ctx);
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    if (ret < 0)
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        return ret;
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    s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
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    if (!s->fifo)
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        return AVERROR(ENOMEM);
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    ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
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    if (ret < 0)
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        return ret;
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    ff_anlmdn_init(&s->dsp);
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    return 0;
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}
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static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
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{
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    AudioNLMeansContext *s = ctx->priv;
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    AVFrame *out = arg;
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    const int S = s->S;
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    const int K = s->K;
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    const int om = s->om;
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    const float *f = (const float *)(s->in->extended_data[ch]) + K;
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    float *cache = (float *)s->cache->extended_data[ch];
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    const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
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    float *dst = (float *)out->extended_data[ch] + s->offset;
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    const float smooth = s->m;
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    for (int i = S; i < s->H + S; i++) {
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        float P = 0.f, Q = 0.f;
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        int v = 0;
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        if (i == S) {
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            for (int j = i - S; j <= i + S; j++) {
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                if (i == j)
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                    continue;
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                cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
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            }
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        } else {
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            s->dsp.compute_cache(cache, f, S, K, i, i - S);
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            s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
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        }
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        for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
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            const float distance = cache[j];
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            unsigned weight_lut_idx;
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            float w;
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            if (distance < 0.f) {
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                cache[j] = 0.f;
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                continue;
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            }
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            w = distance * sw;
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            if (w >= smooth)
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                continue;
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            weight_lut_idx = w * s->pdiff_lut_scale;
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            av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
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            w = s->weight_lut[weight_lut_idx];
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            P += w * f[i - S + j + (j >= S)];
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            Q += w;
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        }
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        P += f[i];
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        Q += 1;
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        switch (om) {
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        case IN_MODE:    dst[i - S] = f[i];           break;
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        case OUT_MODE:   dst[i - S] = P / Q;          break;
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        case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
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        }
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    }
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    return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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    AVFilterContext *ctx = inlink->dst;
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    AVFilterLink *outlink = ctx->outputs[0];
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    AudioNLMeansContext *s = ctx->priv;
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    AVFrame *out = NULL;
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    int available, wanted, ret;
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    if (s->pts == AV_NOPTS_VALUE)
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        s->pts = in->pts;
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    ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
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                              in->nb_samples);
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    av_frame_free(&in);
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    s->offset = 0;
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    available = av_audio_fifo_size(s->fifo);
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    wanted = (available / s->H) * s->H;
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    if (wanted >= s->H && available >= s->N) {
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        out = ff_get_audio_buffer(outlink, wanted);
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        if (!out)
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            return AVERROR(ENOMEM);
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    }
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    while (available >= s->N) {
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        ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
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        if (ret < 0)
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            break;
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        ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);
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        av_audio_fifo_drain(s->fifo, s->H);
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        s->offset += s->H;
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        available -= s->H;
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    }
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    if (out) {
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        out->pts = s->pts;
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        out->nb_samples = s->offset;
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        if (s->eof_left >= 0) {
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            out->nb_samples = FFMIN(s->eof_left, s->offset);
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            s->eof_left -= out->nb_samples;
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        }
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        s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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        return ff_filter_frame(outlink, out);
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    }
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    return ret;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    AudioNLMeansContext *s = ctx->priv;
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    int ret;
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    ret = ff_request_frame(ctx->inputs[0]);
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    if (ret == AVERROR_EOF && s->eof_left != 0) {
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        AVFrame *in;
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        if (s->eof_left < 0)
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            s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
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        if (s->eof_left <= 0)
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            return AVERROR_EOF;
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        in = ff_get_audio_buffer(outlink, s->H);
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        if (!in)
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            return AVERROR(ENOMEM);
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        return filter_frame(ctx->inputs[0], in);
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    }
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    return ret;
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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                           char *res, int res_len, int flags)
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{
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    int ret;
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    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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    if (ret < 0)
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        return ret;
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    ret = config_filter(ctx);
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    if (ret < 0)
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        return ret;
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    return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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    AudioNLMeansContext *s = ctx->priv;
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    av_audio_fifo_free(s->fifo);
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    av_frame_free(&s->in);
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    av_frame_free(&s->cache);
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}
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static const AVFilterPad inputs[] = {
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    {
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        .name         = "default",
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        .type         = AVMEDIA_TYPE_AUDIO,
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        .filter_frame = filter_frame,
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    },
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    { NULL }
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};
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static const AVFilterPad outputs[] = {
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    {
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        .name          = "default",
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        .type          = AVMEDIA_TYPE_AUDIO,
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        .config_props  = config_output,
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        .request_frame = request_frame,
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    },
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    { NULL }
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};
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AVFilter ff_af_anlmdn = {
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    .name          = "anlmdn",
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    .description   = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
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    .query_formats = query_formats,
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    .priv_size     = sizeof(AudioNLMeansContext),
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    .priv_class    = &anlmdn_class,
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    .uninit        = uninit,
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    .inputs        = inputs,
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    .outputs       = outputs,
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    .process_command = process_command,
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    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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                     AVFILTER_FLAG_SLICE_THREADS,
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};