424 lines
11 KiB
C
424 lines
11 KiB
C
/*
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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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};
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