ffmpeg/libavfilter/af_adeclick.c

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/*
* 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 "filters.h"
#include "formats.h"
#include "internal.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 *enabled;
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;
int eof;
AVAudioFifo *efifo;
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[] = {
{ "window", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
{ "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
{ "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
{ "arorder", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 25, AF },
{ "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 25, AF },
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 100, AF },
{ "t", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 100, AF },
{ "burst", "set burst fusion", OFFSET(burst), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, AF },
{ "b", "set burst fusion", OFFSET(burst), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, AF },
{ "method", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
{ "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
{ "add", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "save", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 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)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
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->enabled = ff_get_audio_buffer(inlink, s->window_size);
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 || !s->enabled)
return AVERROR(ENOMEM);
s->efifo = av_audio_fifo_alloc(inlink->format, 1, s->window_size);
if (!s->efifo)
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) {
double *enabled = (double *)s->enabled->extended_data[0];
ret = interpolation(c, src, s->ar_order, c->acoefficients, index,
nb_errors, c->auxiliary, interpolated);
if (ret < 0)
return ret;
av_audio_fifo_peek(s->efifo, (void**)s->enabled->extended_data, s->window_size);
for (j = 0; j < nb_errors; j++) {
if (enabled[index[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)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioDeclickContext *s = ctx->priv;
AVFrame *out = NULL;
int ret = 0, 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)
goto fail;
td.out = out;
ret = ff_filter_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);
av_audio_fifo_drain(s->efifo, s->hop_size);
if (s->samples_left > 0)
out->nb_samples = FFMIN(s->hop_size, s->samples_left);
out->pts = s->pts;
s->pts += av_rescale_q(s->hop_size, (AVRational){1, outlink->sample_rate}, outlink->time_base);
s->detected_errors += detected_errors;
s->nb_samples += out->nb_samples * inlink->channels;
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
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 activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioDeclickContext *s = ctx->priv;
AVFrame *in;
int ret, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->window_size, s->window_size, &in);
if (ret < 0)
return ret;
if (ret > 0) {
double *e = (double *)s->enabled->extended_data[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);
for (int i = 0; i < in->nb_samples; i++)
e[i] = !ctx->is_disabled;
av_audio_fifo_write(s->efifo, (void**)s->enabled->extended_data, in->nb_samples);
av_frame_free(&in);
if (ret < 0)
return ret;
}
if (av_audio_fifo_size(s->fifo) >= s->window_size ||
s->samples_left > 0)
return filter_frame(inlink);
if (av_audio_fifo_size(s->fifo) >= s->window_size) {
ff_filter_set_ready(ctx, 100);
return 0;
}
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF) {
s->eof = 1;
s->samples_left = av_audio_fifo_size(s->fifo) - s->overlap_skip;
ff_filter_set_ready(ctx, 100);
return 0;
}
}
if (s->eof && s->samples_left <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if (!s->eof)
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
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_audio_fifo_free(s->efifo);
av_freep(&s->window_func_lut);
av_frame_free(&s->enabled);
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,
.config_props = config_input,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_adeclick = {
.name = "adeclick",
.description = NULL_IF_CONFIG_SMALL("Remove impulsive noise from input audio."),
.priv_size = sizeof(AudioDeclickContext),
.priv_class = &adeclick_class,
.init = init,
.activate = activate,
.uninit = uninit,
2021-08-12 11:05:31 +00:00
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 10:07:35 +00:00
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
static const AVOption adeclip_options[] = {
{ "window", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
{ "w", "set window size", OFFSET(w), AV_OPT_TYPE_DOUBLE, {.dbl=55}, 10, 100, AF },
{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
{ "o", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_DOUBLE, {.dbl=75}, 50, 95, AF },
{ "arorder", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 0, 25, AF },
{ "a", "set autoregression order", OFFSET(ar), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 0, 25, AF },
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=10}, 1, 100, AF },
{ "t", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=10}, 1, 100, AF },
{ "hsize", "set histogram size", OFFSET(nb_hbins), AV_OPT_TYPE_INT, {.i64=1000}, 100, 9999, AF },
{ "n", "set histogram size", OFFSET(nb_hbins), AV_OPT_TYPE_INT, {.i64=1000}, 100, 9999, AF },
{ "method", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
{ "m", "set overlap method", OFFSET(method), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, AF, "m" },
{ "add", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "a", "overlap-add", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "save", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
{ "s", "overlap-save", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adeclip);
const AVFilter ff_af_adeclip = {
.name = "adeclip",
.description = NULL_IF_CONFIG_SMALL("Remove clipping from input audio."),
.priv_size = sizeof(AudioDeclickContext),
.priv_class = &adeclip_class,
.init = init,
.activate = activate,
.uninit = uninit,
2021-08-12 11:05:31 +00:00
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 10:07:35 +00:00
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};