From 8a1fc95840cb8770e34659803060c00b5b3e733b Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Sat, 19 May 2018 22:06:27 +0200 Subject: avfilter: add anlmdn audio filter Signed-off-by: Paul B Mahol --- libavfilter/af_anlmdn.c | 271 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 271 insertions(+) create mode 100644 libavfilter/af_anlmdn.c (limited to 'libavfilter/af_anlmdn.c') diff --git a/libavfilter/af_anlmdn.c b/libavfilter/af_anlmdn.c new file mode 100644 index 0000000000..62931e37cc --- /dev/null +++ b/libavfilter/af_anlmdn.c @@ -0,0 +1,271 @@ +/* + * 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 + +#include "libavutil/avassert.h" +#include "libavutil/audio_fifo.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +#define SQR(x) ((x) * (x)) + +typedef struct AudioNLMeansContext { + const AVClass *class; + + float a; + int64_t pd; + int64_t rd; + + int K; + int S; + int N; + int H; + + int offset; + AVFrame *in; + AVFrame *cache; + + int64_t pts; + + AVAudioFifo *fifo; + + float (*compute_distance)(const float *f1, const float *f2, int K); +} AudioNLMeansContext; + +#define OFFSET(x) offsetof(AudioNLMeansContext, x) +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption anlmdn_options[] = { + { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=1}, 1, 9999, AF }, + { "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AF }, + { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AF }, + { 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(const float *f1, const float *f2, int K) +{ + float distance = 0.; + + for (int k = -K; k <= K; k++) + distance += SQR(f1[k] - f2[k]); + + return distance; +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioNLMeansContext *s = ctx->priv; + + s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE); + s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE); + + s->pts = AV_NOPTS_VALUE; + s->H = s->K * 2 + 1; + s->N = s->H + (s->K + s->S) * 2; + + av_frame_free(&s->in); + av_frame_free(&s->cache); + s->in = ff_get_audio_buffer(outlink, s->N); + if (!s->in) + return AVERROR(ENOMEM); + + s->cache = ff_get_audio_buffer(outlink, s->S * 2); + if (!s->cache) + return AVERROR(ENOMEM); + + s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N); + if (!s->fifo) + return AVERROR(ENOMEM); + + s->compute_distance = compute_distance_ssd; + + 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 float *f = (const float *)(s->in->extended_data[ch]) + K; + float *cache = (float *)s->cache->extended_data[ch]; + const float sw = 32768.f / s->a; + float *dst = (float *)out->extended_data[ch] + s->offset; + + 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->compute_distance(f + i, f + j, K); + } + } else { + for (int j = i - S; j < i; j++, v++) + cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]); + + for (int j = i + 1; j <= i + S; j++, v++) + cache[v] = cache[v] - SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]); + } + + for (int j = 0; j < v; j++) { + const float distance = cache[j]; + float w; + + av_assert0(distance >= 0.f); + w = expf(-distance * sw); + P += w * f[i - S + j + (j >= S)]; + Q += w; + } + + P += f[i]; + Q += 1; + + dst[i - S] = P / Q; + } + + 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; + s->pts += s->offset; + + return ff_filter_frame(outlink, out); + } + + return ret; +} + +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, + }, + { 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, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; -- cgit v1.2.3