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[79.124.17.100]) by mx.google.com with ESMTP id f3-v6si1345157wrg.265.2018.05.17.10.27.37; Thu, 17 May 2018 10:27:38 -0700 (PDT) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; dkim=neutral (body hash did not verify) header.i=@gmail.com header.s=20161025 header.b=HB/lpnPE; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org; dmarc=fail (p=NONE sp=QUARANTINE dis=NONE) header.from=gmail.com Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 2D08C68A4F4; Thu, 17 May 2018 20:26:56 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail-wm0-f68.google.com (mail-wm0-f68.google.com [74.125.82.68]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 337A668A4D9 for ; Thu, 17 May 2018 20:26:50 +0300 (EEST) Received: by mail-wm0-f68.google.com with SMTP id j5-v6so10486117wme.5 for ; Thu, 17 May 2018 10:27:29 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=from:to:subject:date:message-id; bh=2X7XEakrTovwcCYkQxybhTB9c6EGj5slRKVp+kfAe8o=; b=HB/lpnPEvbOJioB6FxwREP5tg6L8QoVItQNZ9WwknDyuS0WDdtZGz+hVU/WHtMokS+ To3FqmVyY3bGvYRwBSo6FywLKN/opHIdNenWkcMArvvnXobITPGgZYVAukHK50wQNUHf TCvvYq9EVVEqcr574HLL3djLVp0t6MEU4zvxEkCjlzPKq6jWy0zvzNO4S5fLkReA/wnf P3OcKJAnPvWekrO4jh7fECwSzdItBqGXmdIVUs7iehpMl95RJaWBrOOHJtvI3+swQFXU 576W2JadJZvEcyniT6Dit6PYd1t3bulCuiU+mf6g2pOON8etVtIO4xv51e+6eE6e5UCz jK+w== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:from:to:subject:date:message-id; bh=2X7XEakrTovwcCYkQxybhTB9c6EGj5slRKVp+kfAe8o=; b=qSbQVZOibkAHhn3IEtDD6qAnZ/su/CnTYYrYFeWZNq+UvOZWW2wmRtIp6sfhgmi3xN TSNNEPu+pM/vTF2ARvnGPhGIgt8+LP0oJ9yUnZllHbj8J7bucnRtn/vFxgjoH2ZnfrJj 5+XeM9LQx1VeJAEK0mm2hwacupT/cNHbzoByLosgVCzbO2X8FJbFGC8Q+2xTcr9UPHMQ sgCYr8GLSGiQqHxZhQ4CrxWSoxwbDPWrwyjixSLUUfWmFzw+rbWFouShM+VkgNvt1xc0 XSm/Xy6lVIRFCF8GErdjQ5+iWEWJiRTh6SR4stKC79ko+gHq/PfwevltsqL4sw59x8Z7 28fg== X-Gm-Message-State: ALKqPwf+IF2F8g4mPac8y/TuJzHrzYRd3uK4xOCZfaB9gMY+YgwD0Yzd AL0b7NUHMc8AZKJ+Vnz/J7J67Q== X-Received: by 2002:a1c:af4b:: with SMTP id y72-v6mr2347531wme.148.1526577559864; Thu, 17 May 2018 10:19:19 -0700 (PDT) Received: from localhost.localdomain ([94.250.174.60]) by smtp.gmail.com with ESMTPSA id q2-v6sm5953520wrm.26.2018.05.17.10.19.17 for (version=TLS1_2 cipher=ECDHE-RSA-AES128-GCM-SHA256 bits=128/128); Thu, 17 May 2018 10:19:18 -0700 (PDT) From: Paul B Mahol To: ffmpeg-devel@ffmpeg.org Date: Thu, 17 May 2018 19:19:06 +0200 Message-Id: <20180517171906.29056-1-onemda@gmail.com> X-Mailer: git-send-email 2.11.0 Subject: [FFmpeg-devel] [PATCH] avfilter: add declick and declip audio filters X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Signed-off-by: Paul B Mahol --- doc/filters.texi | 95 +++++++ libavfilter/Makefile | 2 + libavfilter/af_declick.c | 699 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 2 + 4 files changed, 798 insertions(+) create mode 100644 libavfilter/af_declick.c diff --git a/doc/filters.texi b/doc/filters.texi index 7646efb918..7ba61135fd 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2576,6 +2576,101 @@ Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping. @end table +@section declick +Remove impulsive noise from input audio. + +Samples detected as impulsive noise are replaced by interpolated samples using +autoregressive modeling. + +@table @option +@item w +Set window size, in milliseconds. Allowed range is from @code{10} to @code{100}. +Default value is @code{55} milliseconds. +This sets size of window which will be processed at once. + +@item o +Set window overlap, in percentage of window size. Allowed range is from @code{50} +to @code{95}. Default value is @code{75} percent. +Setting this to very high value increases impulsive noise removal but makes whole +processs much slower. + +@item a +Set autoregression order, in percentage of window size. Allowed range is from +@code{0} to @code{25}. Default value is @code{2} percent. This option also controls +quality of interpolated samples using neighbour good samples. + +@item t +Set threshold value. Allowed range is from @code{1} to @code{100}. +Default value is @code{2}. +This controls the strength of impulse noise which is going to be removed. + +@item b +Set burst fusion, in percentage of window size. Allowed range is @code{0} to +@code{40}. Default value is @code{10} percent. +This controls between how much samples, which are detected as impulsive noise, +any sample between 2 detected noise samples is considered also as noise sample. + +@item m +Set overlap method. + +It accepts the following values: +@table @option +@item a +Select overlap-add method. Clicks are best removed with this method. +Even not interpolated samples are slightly changed with this method. + +@item s +Select overlap-save method. Less effective method for impulsive noise reduction, +but not interpolated samples remain unchanged. +@end table + +Default value is @code{a}. +@end table + +@section declip +Remove clipped samples from input audio. + +Samples detected as clipped are replaced by interpolated samples using +autoregressive modeling. + +@table @option +@item w +Set window size, in milliseconds. Allowed range is from @code{10} to @code{100}. +Default value is @code{55} milliseconds. +This sets size of window which will be processed at once. + +@item o +Set window overlap, in percentage of window size. Allowed range is from @code{50} +to @code{95}. Default value is @code{75} percent. + +@item a +Set autoregression order, in percentage of window size. Allowed range is from +@code{0} to @code{25}. Default value is @code{8} percent. This option also controls +quality of interpolated samples using neighbour good samples. + +@item t +Set threshold value. Allowed range is from @code{0.2} to @code{1.0}. +Default value is @code{0.98}. +Any sample which absolute value is equal or higher of this value will be +detected as clipped and be replaced with interpolated value. + +@item m +Set overlap method. + +It accepts the following values: +@table @option +@item a +Select overlap-add method. Clips are best removed with this method. +Even not interpolated samples are slightly changed with this method. + +@item s +Select overlap-save method. Less effective method for clip reduction, +but not interpolated samples remain unchanged. +@end table + +Default value is @code{a}. +@end table + @section drmeter Measure audio dynamic range. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index f83a2b30ee..4fa9164429 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -89,6 +89,8 @@ OBJS-$(CONFIG_COMPENSATIONDELAY_FILTER) += af_compensationdelay.o OBJS-$(CONFIG_CROSSFEED_FILTER) += af_crossfeed.o OBJS-$(CONFIG_CRYSTALIZER_FILTER) += af_crystalizer.o OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o +OBJS-$(CONFIG_DECLICK_FILTER) += af_declick.o +OBJS-$(CONFIG_DECLIP_FILTER) += af_declick.o OBJS-$(CONFIG_DRMETER_FILTER) += af_drmeter.o OBJS-$(CONFIG_DYNAUDNORM_FILTER) += af_dynaudnorm.o OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o diff --git a/libavfilter/af_declick.c b/libavfilter/af_declick.c new file mode 100644 index 0000000000..93ca42c975 --- /dev/null +++ b/libavfilter/af_declick.c @@ -0,0 +1,699 @@ +/* + * Copyright (c) 2018 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/audio_fifo.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +typedef struct DeclickChannel { + double *auxiliary; + double *detection; + double *acoefficients; + double *acorrelation; + double *tmp; + double *interpolated; + double *matrix; + int matrix_size; + double *vector; + int vector_size; + double *y; + int y_size; + uint8_t *click; + int *index; +} DeclickChannel; + +typedef struct DeclickContext { + const AVClass *class; + double w; + double overlap; + double threshold; + double clip_threshold; + double ar; + double burst; + int method; + + int is_declip; + int ar_order; + int nb_burst_samples; + int window_size; + int hop_size; + + AVFrame *in; + AVFrame *out; + AVFrame *buffer; + AVFrame *is; + + DeclickChannel *chan; + + int64_t pts; + int nb_channels; + uint64_t nb_samples; + uint64_t detected_errors; + int samples_left; + + AVAudioFifo *fifo; + double *window_func_lut; + + int (*detector)(struct DeclickContext *s, double sigmae, double *detection, + double *acoefficients, uint8_t *click, int *index, + const double *src, double *dst); +} DeclickContext; + +#define OFFSET(x) offsetof(DeclickContext, x) +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption declick_options[] = { + { "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF }, + { "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF }, + { "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 25, AF }, + { "t", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 100, AF }, + { "b", "set burst fusion", OFFSET(burst), AV_OPT_TYPE_DOUBLE, {.dbl=10}, 0, 40, AF }, + { "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" }, + { "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" }, + { "s", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(declick); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + 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 int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + DeclickContext *s = ctx->priv; + int i; + + s->pts = AV_NOPTS_VALUE; + s->window_size = inlink->sample_rate * s->w / 1000.; + if (s->window_size < 100) + return AVERROR(EINVAL); + s->ar_order = FFMAX(s->window_size * s->ar / 100., 1); + s->nb_burst_samples = s->window_size * s->burst / 1000.; + s->hop_size = s->window_size * (1. - (s->overlap / 100.)); + if (s->hop_size < 1) + return AVERROR(EINVAL); + + s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size); + if (!s->fifo) + return AVERROR(ENOMEM); + + s->window_func_lut = av_realloc_f(s->window_func_lut, s->window_size, + sizeof(*s->window_func_lut)); + if (!s->window_func_lut) + return AVERROR(ENOMEM); + for (i = 0; i < s->window_size; i++) + s->window_func_lut[i] = sin(M_PI * i / s->window_size) * (1. - (s->overlap / 100.)) * M_PI_2; + + av_frame_free(&s->in); + av_frame_free(&s->out); + s->in = ff_get_audio_buffer(inlink, s->window_size); + s->out = ff_get_audio_buffer(inlink, s->window_size); + s->buffer = ff_get_audio_buffer(inlink, s->window_size * 2); + s->is = ff_get_audio_buffer(inlink, s->window_size); + if (!s->in || !s->out || !s->buffer || !s->is) + return AVERROR(ENOMEM); + + s->nb_channels = inlink->channels; + s->chan = av_calloc(inlink->channels, sizeof(*s->chan)); + if (!s->chan) + return AVERROR(ENOMEM); + + for (i = 0; i < inlink->channels; i++) { + DeclickChannel *c = &s->chan[i]; + + c->detection = av_calloc(s->window_size, sizeof(*c->detection)); + c->auxiliary = av_calloc(s->ar_order + 1, sizeof(*c->auxiliary)); + c->acoefficients = av_calloc(s->ar_order + 1, sizeof(*c->acoefficients)); + c->acorrelation = av_calloc(s->ar_order + 1, sizeof(*c->acorrelation)); + c->tmp = av_calloc(s->ar_order, sizeof(*c->tmp)); + c->click = av_calloc(s->window_size, sizeof(*c->click)); + c->index = av_calloc(s->window_size, sizeof(*c->index)); + c->interpolated = av_calloc(s->window_size, sizeof(*c->interpolated)); + if (!c->auxiliary || !c->acoefficients || !c->detection || !c->click || + !c->index || !c->interpolated || !c->acorrelation || !c->tmp) + return AVERROR(ENOMEM); + } + + return 0; +} + +static void autocorrelation(const double *input, int order, int size, double *output, double scale) +{ + int i, j; + + for (i = 0; i <= order; i++) { + double value = 0.; + + for (j = i; j < size; j++) + value += input[j] * input[j - i]; + + output[i] = value * scale; + } +} + +static double autoregression(const double *samples, int ar_order, int nb_samples, double *k, double *r, double *a) +{ + double alpha; + int i, j; + + memset(a, 0, ar_order * sizeof(*a)); + + autocorrelation(samples, ar_order, nb_samples, r, 1. / nb_samples); + + /* Levinson-Durbin algorithm */ + k[0] = a[0] = -r[1] / r[0]; + alpha = r[0] * (1. - k[0] * k[0]); + for (i = 1; i < ar_order; i++) { + double epsilon = 0.; + + for (j = 0; j < i; j++) + epsilon += a[j] * r[i - j]; + epsilon += r[i + 1]; + + k[i] = -epsilon / alpha; + alpha *= (1. - k[i] * k[i]); + for (j = i - 1; j >= 0; j--) + k[j] = a[j] + k[i] * a[i - j - 1]; + for (j = 0; j <= i; j++) + a[j] = k[j]; + } + + k[0] = 1.; + for (i = 0; i < ar_order; i++) + k[i + 1] = a[i]; + + return sqrt(alpha); +} + +static int isfinite_array(double *samples, int nb_samples) +{ + int i; + + for (i = 0; i < nb_samples; i++) + if (!isfinite(samples[i])) + return 0; + + return 1; +} + +static int find_index(int *index, int value, int size) +{ + int i, start, end; + + if ((value < index[0]) || (value > index[size - 1])) + return 1; + + i = start = 0; + end = size - 1; + + while (start <= end) { + i = (end + start) / 2; + if (index[i] == value) + return 0; + if (value < index[i]) + end = i - 1; + if (value > index[i]) + start = i + 1; + } + + return 1; +} + +static int factorization(double *matrix, int n) +{ + int i, j, k; + + for (i = 0; i < n; i++) { + const int in = i * n; + double value; + + value = matrix[in + i]; + for (j = 0; j < i; j++) + value -= matrix[j * n + j] * matrix[in + j] * matrix[in + j]; + + if (value == 0.) { + return -1; + } + + matrix[in + i] = value; + for (j = i + 1; j < n; j++) { + const int jn = j * n; + double x; + + x = matrix[jn + i]; + for (k = 0; k < i; k++) + x -= matrix[k * n + k] * matrix[in + k] * matrix[jn + k]; + matrix[jn + i] = x / matrix[in + i]; + } + } + + return 0; +} + +static int do_interpolation(DeclickChannel *c, double *matrix, double *vector, int n, double *out) +{ + int i, j, ret; + double *y; + + ret = factorization(matrix, n); + if (ret < 0) + return ret; + + av_fast_malloc(&c->y, &c->y_size, n * sizeof(*c->y)); + y = c->y; + if (!y) + return AVERROR(ENOMEM); + + for (i = 0; i < n; i++) { + const int in = i * n; + double value; + + value = vector[i]; + for (j = 0; j < i; j++) + value -= matrix[in + j] * y[j]; + y[i] = value; + } + + for (i = n - 1; i >= 0; i--) { + out[i] = y[i] / matrix[i * n + i]; + for (j = i + 1; j < n; j++) + out[i] -= matrix[j * n + i] * out[j]; + } + + return 0; +} + +static int interpolation(DeclickChannel *c, const double *src, int ar_order, + double *acoefficients, int *index, int nb_errors, + double *auxiliary, double *interpolated) +{ + double *vector, *matrix; + int i, j; + + av_fast_malloc(&c->matrix, &c->matrix_size, nb_errors * nb_errors * sizeof(*c->matrix)); + matrix = c->matrix; + if (!matrix) + return AVERROR(ENOMEM); + + av_fast_malloc(&c->vector, &c->vector_size, nb_errors * sizeof(*c->vector)); + vector = c->vector; + if (!vector) + return AVERROR(ENOMEM); + + autocorrelation(acoefficients, ar_order, ar_order + 1, auxiliary, 1.); + + for (i = 0; i < nb_errors; i++) { + const int im = i * nb_errors; + + for (j = i; j < nb_errors; j++) { + if (abs(index[j] - index[i]) <= ar_order) { + matrix[j * nb_errors + i] = matrix[im + j] = auxiliary[abs(index[j] - index[i])]; + } else { + matrix[j * nb_errors + i] = matrix[im + j] = 0; + } + } + } + + for (i = 0; i < nb_errors; i++) { + double value = 0.; + + for (j = -ar_order; j <= ar_order; j++) + if (find_index(index, index[i] - j, nb_errors)) + value -= src[index[i] - j] * auxiliary[abs(j)]; + + vector[i] = value; + } + + return do_interpolation(c, matrix, vector, nb_errors, interpolated); +} + +static int detect_clips(DeclickContext *s, double unused0, double *unused1, double *unused2, + uint8_t *clips, int *index, + const double *src, double *dst) +{ + const double threshold = s->clip_threshold; + int i, nb_clips = 0; + + for (i = 0; i < s->window_size; i++) { + clips[i] = fabs(src[i]) >= threshold; + dst[i] = src[i]; + } + + memset(clips, 0, s->ar_order * sizeof(*clips)); + memset(clips + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*clips)); + + for (i = s->ar_order; i < s->window_size - s->ar_order; i++) + if (clips[i]) + index[nb_clips++] = i; + + return nb_clips; +} + +static int detect_clicks(DeclickContext *s, double sigmae, double *detection, double *acoefficients, + uint8_t *click, int *index, + const double *src, double *dst) +{ + const double threshold = s->threshold; + int i, j, nb_clicks = 0, prev = -1; + + memset(detection, 0, s->window_size * sizeof(*detection)); + + for (i = s->ar_order; i < s->window_size; i++) { + for (j = 0; j <= s->ar_order; j++) { + detection[i] += acoefficients[j] * src[i - j]; + } + } + + for (i = 0; i < s->window_size; i++) { + click[i] = fabs(detection[i]) > sigmae * threshold; + dst[i] = src[i]; + } + + for (i = 0; i < s->window_size; i++) { + if (!click[i]) + continue; + + if (prev >= 0 && (i > prev + 1) && (i <= s->nb_burst_samples + prev)) + for (j = prev + 1; j < i; j++) + click[j] = 1; + prev = i; + } + + memset(click, 0, s->ar_order * sizeof(*click)); + memset(click + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*click)); + + for (i = s->ar_order; i < s->window_size - s->ar_order; i++) + if (click[i]) + index[nb_clicks++] = i; + + return nb_clicks; +} + +typedef struct ThreadData { + AVFrame *out; +} ThreadData; + +static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs) +{ + DeclickContext *s = ctx->priv; + ThreadData *td = arg; + AVFrame *out = td->out; + const double *src = (const double *)s->in->extended_data[ch]; + double *is = (double *)s->is->extended_data[ch]; + double *dst = (double *)s->out->extended_data[ch]; + double *ptr = (double *)out->extended_data[ch]; + double *buf = (double *)s->buffer->extended_data[ch]; + const double *w = s->window_func_lut; + DeclickChannel *c = &s->chan[ch]; + double sigmae; + int j, ret; + + sigmae = autoregression(src, s->ar_order, s->window_size, c->acoefficients, c->acorrelation, c->tmp); + + if (isfinite_array(c->acoefficients, s->ar_order + 1)) { + double *interpolated = c->interpolated; + int *index = c->index; + int nb_errors; + + nb_errors = s->detector(s, sigmae, c->detection, c->acoefficients, + c->click, index, src, dst); + if (nb_errors > 0) { + ret = interpolation(c, src, s->ar_order, c->acoefficients, index, + nb_errors, c->auxiliary, interpolated); + if (ret < 0) + return ret; + + for (j = 0; j < nb_errors; j++) { + dst[index[j]] = interpolated[j]; + is[index[j]] = 1; + } + } + } else { + memcpy(dst, src, s->window_size * sizeof(*dst)); + } + + if (s->method == 0) { + for (j = 0; j < s->window_size; j++) + buf[j] += dst[j] * w[j]; + } else { + for (j = 0; j < s->window_size; j++) + buf[j] = dst[j]; + } + for (j = 0; j < s->hop_size; j++) + ptr[j] = buf[j]; + + memmove(buf, buf + s->hop_size, (s->window_size * 2 - s->hop_size) * sizeof(*buf)); + memmove(is, is + s->hop_size, (s->window_size - s->hop_size) * sizeof(*is)); + memset(buf + s->window_size * 2 - s->hop_size, 0, s->hop_size * sizeof(*buf)); + memset(is + s->window_size - s->hop_size, 0, s->hop_size * sizeof(*is)); + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + DeclickContext *s = ctx->priv; + AVFrame *out = NULL; + int ret = 0; + + 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); + + while (av_audio_fifo_size(s->fifo) >= s->window_size) { + int j, ch, detected_errors = 0; + ThreadData td; + + out = ff_get_audio_buffer(outlink, s->hop_size); + if (!out) + return AVERROR(ENOMEM); + + ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, + s->window_size); + if (ret < 0) + break; + + td.out = out; + ret = ctx->internal->execute(ctx, filter_channel, &td, NULL, inlink->channels); + if (ret < 0) + goto fail; + + for (ch = 0; ch < s->in->channels; ch++) { + double *is = (double *)s->is->extended_data[ch]; + + for (j = 0; j < s->hop_size; j++) { + if (is[j]) + detected_errors++; + } + } + + av_audio_fifo_drain(s->fifo, s->hop_size); + + out->pts = s->pts; + s->pts += s->hop_size; + + s->detected_errors += detected_errors; + s->nb_samples += s->hop_size * inlink->channels; + + ret = ff_filter_frame(outlink, out); + if (ret < 0) + break; + } + +fail: + if (ret < 0) + av_frame_free(&out); + return ret; +} + +static int request_frame(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + DeclickContext *s = ctx->priv; + int ret = 0; + + ret = ff_request_frame(ctx->inputs[0]); + + if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0) { + AVFrame *in; + + if (!s->samples_left) + s->samples_left = av_audio_fifo_size(s->fifo); + + in = ff_get_audio_buffer(outlink, s->window_size); + if (!in) + return AVERROR(ENOMEM); + ret = filter_frame(ctx->inputs[0], in); + if (s->samples_left) { + s->samples_left -= s->hop_size; + if (s->samples_left <= 0) + av_audio_fifo_drain(s->fifo, s->window_size); + } + } + + return ret; +} + +static av_cold int init(AVFilterContext *ctx) +{ + DeclickContext *s = ctx->priv; + + s->is_declip = !strcmp(ctx->filter->name, "declip"); + if (s->is_declip) { + s->detector = detect_clips; + } else { + s->detector = detect_clicks; + } + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + DeclickContext *s = ctx->priv; + int i; + + av_log(ctx, AV_LOG_INFO, "Detected %s in %"PRId64" of %"PRId64" samples (%g%%).\n", + s->is_declip ? "clips" : "clicks", s->detected_errors, + s->nb_samples, 100. * s->detected_errors / s->nb_samples); + + av_audio_fifo_free(s->fifo); + av_freep(&s->window_func_lut); + av_frame_free(&s->in); + av_frame_free(&s->out); + av_frame_free(&s->buffer); + av_frame_free(&s->is); + + if (s->chan) { + for (i = 0; i < s->nb_channels; i++) { + DeclickChannel *c = &s->chan[i]; + + av_freep(&c->detection); + av_freep(&c->auxiliary); + av_freep(&c->acoefficients); + av_freep(&c->acorrelation); + av_freep(&c->tmp); + av_freep(&c->click); + av_freep(&c->index); + av_freep(&c->interpolated); + av_freep(&c->matrix); + c->matrix_size = 0; + av_freep(&c->vector); + c->vector_size = 0; + av_freep(&c->y); + c->y_size = 0; + } + } + av_freep(&s->chan); + s->nb_channels = 0; +} + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .request_frame = request_frame, + }, + { NULL } +}; + +AVFilter ff_af_declick = { + .name = "declick", + .description = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."), + .query_formats = query_formats, + .priv_size = sizeof(DeclickContext), + .priv_class = &declick_class, + .init = init, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; + +static const AVOption declip_options[] = { + { "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF }, + { "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF }, + { "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 0, 25, AF }, + { "t", "set threshold", OFFSET(clip_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.98}, 0.2, 1., AF }, + { "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" }, + { "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" }, + { "s", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(declip); + +AVFilter ff_af_declip = { + .name = "declip", + .description = NULL_IF_CONFIG_SMALL("Remove clipping from input audio."), + .query_formats = query_formats, + .priv_size = sizeof(DeclickContext), + .priv_class = &declip_class, + .init = init, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, + .flags = AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 099c19157b..bca217e616 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -84,6 +84,8 @@ extern AVFilter ff_af_compensationdelay; extern AVFilter ff_af_crossfeed; extern AVFilter ff_af_crystalizer; extern AVFilter ff_af_dcshift; +extern AVFilter ff_af_declick; +extern AVFilter ff_af_declip; extern AVFilter ff_af_drmeter; extern AVFilter ff_af_dynaudnorm; extern AVFilter ff_af_earwax;