[FFmpeg-cvslog] avfilter: add adeclick and adeclip audio filters
Paul B Mahol
git at videolan.org
Sat Jun 2 14:11:11 EEST 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Mon Jan 8 15:02:59 2018 +0100| [e28b1fa6e99c9a50d1b647c5418e5f75a3f957e4] | committer: Paul B Mahol
avfilter: add adeclick and adeclip audio filters
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e28b1fa6e99c9a50d1b647c5418e5f75a3f957e4
---
Changelog | 2 +
doc/filters.texi | 96 ++++++
libavfilter/Makefile | 2 +
libavfilter/af_adeclick.c | 753 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 2 +
libavfilter/version.h | 2 +-
6 files changed, 856 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index df2024fb59..9a96e8db6d 100644
--- a/Changelog
+++ b/Changelog
@@ -11,6 +11,8 @@ version <next>:
- support mbedTLS based TLS
- DNN inference interface
- Reimplemented SRCNN filter using DNN inference interface
+- adeclick filter
+- adeclip filter
version 4.0:
diff --git a/doc/filters.texi b/doc/filters.texi
index fb131670c7..cbb06afbfd 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -551,6 +551,102 @@ Set LFO range.
Set LFO rate.
@end table
+ at section adeclick
+Remove impulsive noise from input audio.
+
+Samples detected as impulsive noise are replaced by interpolated samples using
+autoregressive modelling.
+
+ at table @option
+ at item w
+Set window size, in milliseconds. Allowed range is from @code{10} to
+ at code{100}. Default value is @code{55} milliseconds.
+This sets size of window which will be processed at once.
+
+ at item o
+Set window overlap, in percentage of window size. Allowed range is from
+ at code{50} to @code{95}. Default value is @code{75} percent.
+Setting this to a very high value increases impulsive noise removal but makes
+whole process much slower.
+
+ at item a
+Set autoregression order, in percentage of window size. Allowed range is from
+ at code{0} to @code{25}. Default value is @code{2} percent. This option also
+controls quality of interpolated samples using neighbour good samples.
+
+ at item t
+Set threshold value. Allowed range is from @code{1} to @code{100}.
+Default value is @code{2}.
+This controls the strength of impulsive noise which is going to be removed.
+The lower value, the more samples will be detected as impulsive noise.
+
+ at item b
+Set burst fusion, in percentage of window size. Allowed range is @code{0} to
+ at code{10}. Default value is @code{2}.
+If any two samples deteced as noise are spaced less than this value then any
+sample inbetween those two samples will be also detected as noise.
+
+ at item m
+Set overlap method.
+
+It accepts the following values:
+ at table @option
+ at item a
+Select overlap-add method. Even not interpolated samples are slightly
+changed with this method.
+
+ at item s
+Select overlap-save method. Not interpolated samples remain unchanged.
+ at end table
+
+Default value is @code{a}.
+ at end table
+
+ at section adeclip
+Remove clipped samples from input audio.
+
+Samples detected as clipped are replaced by interpolated samples using
+autoregressive modelling.
+
+ at table @option
+ at 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.
+
+ at 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.
+
+ at item a
+Set autoregression order, in percentage of window size. Allowed range is from
+ at code{0} to @code{25}. Default value is @code{8} percent. This option also controls
+quality of interpolated samples using neighbour good samples.
+
+ at item t
+Set threshold value. Allowed range is from @code{1} to @code{100}.
+Default value is @code{10}. Higher values make clip detection less aggressive.
+
+ at item n
+Set size of histogram used to detect clips. Allowed range is from @code{100} to @code{9999}.
+Default value is @code{1000}. Higher values make clip detection less aggressive.
+
+ at item m
+Set overlap method.
+
+It accepts the following values:
+ at table @option
+ at item a
+Select overlap-add method. Even not interpolated samples are slightly changed
+with this method.
+
+ at item s
+Select overlap-save method. Not interpolated samples remain unchanged.
+ at end table
+
+Default value is @code{a}.
+ at end table
+
@section adelay
Delay one or more audio channels.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 3201cbeacf..5bacd5b621 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -36,6 +36,8 @@ OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o
OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o
+OBJS-$(CONFIG_ADECLICK_FILTER) += af_adeclick.o
+OBJS-$(CONFIG_ADECLIP_FILTER) += af_adeclick.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
diff --git a/libavfilter/af_adeclick.c b/libavfilter/af_adeclick.c
new file mode 100644
index 0000000000..bf0b7cb408
--- /dev/null
+++ b/libavfilter/af_adeclick.c
@@ -0,0 +1,753 @@
+/*
+ * 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;
+ unsigned *histogram;
+ int histogram_size;
+} DeclickChannel;
+
+typedef struct AudioDeclickContext {
+ const AVClass *class;
+
+ double w;
+ double overlap;
+ double threshold;
+ double ar;
+ double burst;
+ int method;
+ int nb_hbins;
+
+ int is_declip;
+ int ar_order;
+ int nb_burst_samples;
+ int window_size;
+ int hop_size;
+ int overlap_skip;
+
+ 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 AudioDeclickContext *s, DeclickChannel *c,
+ double sigmae, double *detection,
+ double *acoefficients, uint8_t *click, int *index,
+ const double *src, double *dst);
+} AudioDeclickContext;
+
+#define OFFSET(x) offsetof(AudioDeclickContext, x)
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adeclick_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=2}, 0, 10, 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(adeclick);
+
+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;
+ AudioDeclickContext *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->window_func_lut = av_calloc(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);
+ av_frame_free(&s->buffer);
+ av_frame_free(&s->is);
+ 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->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->window_size);
+ if (!s->fifo)
+ return AVERROR(ENOMEM);
+ s->overlap_skip = s->method ? (s->window_size - s->hop_size) / 2 : 0;
+ if (s->overlap_skip > 0) {
+ av_audio_fifo_write(s->fifo, (void **)s->in->extended_data,
+ s->overlap_skip);
+ }
+
+ 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 = 1; i <= ar_order; i++)
+ k[i] = a[i - 1];
+
+ 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(AudioDeclickContext *s, DeclickChannel *c,
+ double unused0,
+ double *unused1, double *unused2,
+ uint8_t *clip, int *index,
+ const double *src, double *dst)
+{
+ const double threshold = s->threshold;
+ double max_amplitude = 0;
+ unsigned *histogram;
+ int i, nb_clips = 0;
+
+ av_fast_malloc(&c->histogram, &c->histogram_size, s->nb_hbins * sizeof(*c->histogram));
+ if (!c->histogram)
+ return AVERROR(ENOMEM);
+ histogram = c->histogram;
+ memset(histogram, 0, sizeof(*histogram) * s->nb_hbins);
+
+ for (i = 0; i < s->window_size; i++) {
+ const unsigned index = fmin(fabs(src[i]), 1) * (s->nb_hbins - 1);
+
+ histogram[index]++;
+ dst[i] = src[i];
+ clip[i] = 0;
+ }
+
+ for (i = s->nb_hbins - 1; i > 1; i--) {
+ if (histogram[i]) {
+ if (histogram[i] / (double)FFMAX(histogram[i - 1], 1) > threshold) {
+ max_amplitude = i / (double)s->nb_hbins;
+ }
+ break;
+ }
+ }
+
+ if (max_amplitude > 0.) {
+ for (i = 0; i < s->window_size; i++) {
+ clip[i] = fabs(src[i]) >= max_amplitude;
+ }
+ }
+
+ memset(clip, 0, s->ar_order * sizeof(*clip));
+ memset(clip + (s->window_size - s->ar_order), 0, s->ar_order * sizeof(*clip));
+
+ for (i = s->ar_order; i < s->window_size - s->ar_order; i++)
+ if (clip[i])
+ index[nb_clips++] = i;
+
+ return nb_clips;
+}
+
+static int detect_clicks(AudioDeclickContext *s, DeclickChannel *c,
+ 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)
+{
+ AudioDeclickContext *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, c, 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 {
+ const int skip = s->overlap_skip;
+
+ for (j = 0; j < s->hop_size; j++)
+ buf[j] = dst[skip + 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];
+ AudioDeclickContext *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);
+
+ if (s->samples_left > 0)
+ out->nb_samples = FFMIN(s->hop_size, s->samples_left);
+
+ out->pts = s->pts;
+ s->pts += s->hop_size;
+
+ s->detected_errors += detected_errors;
+ s->nb_samples += out->nb_samples * inlink->channels;
+
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ break;
+
+ if (s->samples_left > 0) {
+ s->samples_left -= s->hop_size;
+ if (s->samples_left <= 0)
+ av_audio_fifo_drain(s->fifo, av_audio_fifo_size(s->fifo));
+ }
+ }
+
+fail:
+ if (ret < 0)
+ av_frame_free(&out);
+ return ret;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioDeclickContext *s = ctx->priv;
+ int ret = 0;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0) {
+ if (!s->samples_left)
+ s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip;
+
+ if (s->samples_left > 0) {
+ AVFrame *in = ff_get_audio_buffer(outlink, s->window_size - s->samples_left);
+ if (!in)
+ return AVERROR(ENOMEM);
+ ret = filter_frame(ctx->inputs[0], in);
+ }
+ }
+
+ return ret;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioDeclickContext *s = ctx->priv;
+
+ s->is_declip = !strcmp(ctx->filter->name, "adeclip");
+ if (s->is_declip) {
+ s->detector = detect_clips;
+ } else {
+ s->detector = detect_clicks;
+ }
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioDeclickContext *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->histogram);
+ c->histogram_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_adeclick = {
+ .name = "adeclick",
+ .description = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioDeclickContext),
+ .priv_class = &adeclick_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+};
+
+static const AVOption adeclip_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(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=10}, 1, 100, AF },
+ { "n", "set histogram size", OFFSET(nb_hbins), AV_OPT_TYPE_INT, {.i64=1000}, 100, 9999, 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(adeclip);
+
+AVFilter ff_af_adeclip = {
+ .name = "adeclip",
+ .description = NULL_IF_CONFIG_SMALL("Remove clipping from input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioDeclickContext),
+ .priv_class = &adeclip_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 b44093d21b..f2d27d2424 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -29,6 +29,8 @@ extern AVFilter ff_af_acontrast;
extern AVFilter ff_af_acopy;
extern AVFilter ff_af_acrossfade;
extern AVFilter ff_af_acrusher;
+extern AVFilter ff_af_adeclick;
+extern AVFilter ff_af_adeclip;
extern AVFilter ff_af_adelay;
extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index c32afce3e9..a7be7e64af 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 24
+#define LIBAVFILTER_VERSION_MINOR 25
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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