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avfilter: add audio psychoacoustic clipper

master
Paul B Mahol 3 months ago
parent
commit
eeab62ad2d
  1. 1
      Changelog
  2. 36
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 679
      libavfilter/af_apsyclip.c
  5. 1
      libavfilter/allfilters.c
  6. 2
      libavfilter/version.h

1
Changelog

@ -19,6 +19,7 @@ version <next>:
- swscale slice threading
- MSN Siren decoder
- scharr video filter
- apsyclip audio filter
version 4.4:

36
doc/filters.texi

@ -2363,6 +2363,42 @@ Default value is 8.
This filter supports the all above options as @ref{commands}.
@section apsyclip
Apply Psychoacoustic clipper to input audio stream.
The filter accepts the following options:
@table @option
@item level_in
Set input gain. By default it is 1. Range is [0.015625 - 64].
@item level_out
Set output gain. By default it is 1. Range is [0.015625 - 64].
@item clip
Set the clipping start value. Default value is 0dBFS or 1.
@item diff
Output only difference samples, useful to hear introduced distortions.
By default is disabled.
@item adaptive
Set strenght of adaptive distortion applied. Default value is 0.5.
Allowed range is from 0 to 1.
@item iterations
Set number of iterations of psychoacoustic clipper.
Allowed range is from 1 to 20. Default value is 10.
@item level
Auto level output signal. Default is disabled.
This normalizes audio back to 0dBFS if enabled.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.

1
libavfilter/Makefile

@ -74,6 +74,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o

679
libavfilter/af_apsyclip.c

@ -0,0 +1,679 @@
/*
* Copyright (c) 2014 - 2021 Jason Jang
* Copyright (c) 2021 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/opt.h"
#include "libavutil/tx.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
typedef struct AudioPsyClipContext {
const AVClass *class;
double level_in;
double level_out;
double clip_level;
double adaptive;
int auto_level;
int diff_only;
int iterations;
char *protections_str;
double *protections;
int num_psy_bins;
int fft_size;
int overlap;
int channels;
int spread_table_rows;
int *spread_table_index;
int (*spread_table_range)[2];
float *window, *inv_window, *spread_table, *margin_curve;
AVFrame *in;
AVFrame *in_buffer;
AVFrame *in_frame;
AVFrame *out_dist_frame;
AVFrame *windowed_frame;
AVFrame *clipping_delta;
AVFrame *spectrum_buf;
AVFrame *mask_curve;
AVTXContext **tx_ctx;
av_tx_fn tx_fn;
AVTXContext **itx_ctx;
av_tx_fn itx_fn;
} AudioPsyClipContext;
#define OFFSET(x) offsetof(AudioPsyClipContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption apsyclip_options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
{ "clip", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, FLAGS },
{ "diff", "enable difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ "adaptive", "set adaptive distortion", OFFSET(adaptive), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
{ "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 1, 20, FLAGS },
{ "level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{NULL}
};
AVFILTER_DEFINE_CLASS(apsyclip);
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
int ret;
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static void generate_hann_window(float *window, float *inv_window, int size)
{
for (int i = 0; i < size; i++) {
float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
window[i] = value;
// 1/window to calculate unwindowed peak.
inv_window[i] = value > 0.01f ? 1.f / value : 0.f;
}
}
static void set_margin_curve(AudioPsyClipContext *s,
const int (*points)[2], int num_points, int sample_rate)
{
int j = 0;
s->margin_curve[0] = points[0][1];
for (int i = 0; i < num_points - 1; i++) {
while (j < s->fft_size / 2 + 1 && j * sample_rate / s->fft_size < points[i + 1][0]) {
// linearly interpolate between points
int binHz = j * sample_rate / s->fft_size;
s->margin_curve[j] = points[i][1] + (binHz - points[i][0]) * (points[i + 1][1] - points[i][1]) / (points[i + 1][0] - points[i][0]);
j++;
}
}
// handle bins after the last point
while (j < s->fft_size / 2 + 1) {
s->margin_curve[j] = points[num_points - 1][1];
j++;
}
// convert margin curve to linear amplitude scale
for (j = 0; j < s->fft_size / 2 + 1; j++)
s->margin_curve[j] = powf(10.f, s->margin_curve[j] / 20.f);
}
static void generate_spread_table(AudioPsyClipContext *s)
{
// Calculate tent-shape function in log-log scale.
// As an optimization, only consider bins close to "bin"
// This reduces the number of multiplications needed in calculate_mask_curve
// The masking contribution at faraway bins is negligeable
// Another optimization to save memory and speed up the calculation of the
// spread table is to calculate and store only 2 spread functions per
// octave, and reuse the same spread function for multiple bins.
int table_index = 0;
int bin = 0;
int increment = 1;
while (bin < s->num_psy_bins) {
float sum = 0;
int base_idx = table_index * s->num_psy_bins;
int start_bin = bin * 3 / 4;
int end_bin = FFMIN(s->num_psy_bins, ((bin + 1) * 4 + 2) / 3);
int next_bin;
for (int j = start_bin; j < end_bin; j++) {
// add 0.5 so i=0 doesn't get log(0)
float rel_idx_log = FFABS(logf((j + 0.5f) / (bin + 0.5f)));
float value;
if (j >= bin) {
// mask up
value = expf(-rel_idx_log * 40.f);
} else {
// mask down
value = expf(-rel_idx_log * 80.f);
}
// the spreading function is centred in the row
sum += value;
s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] = value;
}
// now normalize it
for (int j = start_bin; j < end_bin; j++) {
s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] /= sum;
}
s->spread_table_range[table_index][0] = start_bin - bin;
s->spread_table_range[table_index][1] = end_bin - bin;
if (bin <= 1) {
next_bin = bin + 1;
} else {
if ((bin & (bin - 1)) == 0) {
// power of 2
increment = bin / 2;
}
next_bin = bin + increment;
}
// set bins between "bin" and "next_bin" to use this table_index
for (int i = bin; i < next_bin; i++)
s->spread_table_index[i] = table_index;
bin = next_bin;
table_index++;
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioPsyClipContext *s = ctx->priv;
static const int points[][2] = { {0,14}, {125,14}, {250,16}, {500,18}, {1000,20}, {2000,20}, {4000,20}, {8000,15}, {16000,5}, {20000,-10} };
static const int num_points = 10;
float scale;
int ret;
s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
s->overlap = s->fft_size / 4;
// The psy masking calculation is O(n^2),
// so skip it for frequencies not covered by base sampling rantes (i.e. 44k)
if (inlink->sample_rate <= 50000) {
s->num_psy_bins = s->fft_size / 2;
} else if (inlink->sample_rate <= 100000) {
s->num_psy_bins = s->fft_size / 4;
} else {
s->num_psy_bins = s->fft_size / 8;
}
s->window = av_calloc(s->fft_size, sizeof(*s->window));
s->inv_window = av_calloc(s->fft_size, sizeof(*s->inv_window));
if (!s->window || !s->inv_window)
return AVERROR(ENOMEM);
s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->clipping_delta = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
s->mask_curve = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
if (!s->in_buffer || !s->in_frame ||
!s->out_dist_frame || !s->windowed_frame ||
!s->clipping_delta || !s->spectrum_buf || !s->mask_curve)
return AVERROR(ENOMEM);
generate_hann_window(s->window, s->inv_window, s->fft_size);
s->margin_curve = av_calloc(s->fft_size / 2 + 1, sizeof(*s->margin_curve));
if (!s->margin_curve)
return AVERROR(ENOMEM);
s->spread_table_rows = av_log2(s->num_psy_bins) * 2;
s->spread_table = av_calloc(s->spread_table_rows * s->num_psy_bins, sizeof(*s->spread_table));
if (!s->spread_table)
return AVERROR(ENOMEM);
s->spread_table_range = av_calloc(s->spread_table_rows * 2, sizeof(*s->spread_table_range));
if (!s->spread_table_range)
return AVERROR(ENOMEM);
s->spread_table_index = av_calloc(s->num_psy_bins, sizeof(*s->spread_table_index));
if (!s->spread_table_index)
return AVERROR(ENOMEM);
set_margin_curve(s, points, num_points, inlink->sample_rate);
generate_spread_table(s);
s->channels = inlink->channels;
s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
if (!s->tx_ctx || !s->itx_ctx)
return AVERROR(ENOMEM);
for (int ch = 0; ch < s->channels; ch++) {
ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_FFT, 1, s->fft_size, &scale, 0);
if (ret < 0)
return ret;
}
return 0;
}
static void apply_window(AudioPsyClipContext *s,
const float *in_frame, float *out_frame, const int add_to_out_frame)
{
const float *window = s->window;
for (int i = 0; i < s->fft_size; i++) {
if (add_to_out_frame) {
out_frame[i] += in_frame[i] * window[i];
} else {
out_frame[i] = in_frame[i] * window[i];
}
}
}
static void calculate_mask_curve(AudioPsyClipContext *s,
const float *spectrum, float *mask_curve)
{
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] = 0;
for (int i = 0; i < s->num_psy_bins; i++) {
float magnitude;
int table_idx;
int range[2];
if (i == 0) {
magnitude = FFABS(spectrum[0]);
} else if (i == s->fft_size / 2) {
magnitude = FFABS(spectrum[1]);
} else {
// although the negative frequencies are omitted because they are redundant,
// the magnitude of the positive frequencies are not doubled.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
}
table_idx = s->spread_table_index[i];
range[0] = s->spread_table_range[table_idx][0];
range[1] = s->spread_table_range[table_idx][1];
int base_idx = table_idx * s->num_psy_bins;
int start_bin = FFMAX(0, i + range[0]);
int end_bin = FFMIN(s->num_psy_bins, i + range[1]);
for (int j = start_bin; j < end_bin; j++)
mask_curve[j] += s->spread_table[base_idx + s->num_psy_bins / 2 + j - i] * magnitude;
}
// for ultrasonic frequencies, skip the O(n^2) spread calculation and just copy the magnitude
for (int i = s->num_psy_bins; i < s->fft_size / 2 + 1; i++) {
float magnitude;
if (i == s->fft_size / 2) {
magnitude = FFABS(spectrum[1]);
} else {
// although the negative frequencies are omitted because they are redundant,
// the magnitude of the positive frequencies are not doubled.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
}
mask_curve[i] = magnitude;
}
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] = mask_curve[i] / s->margin_curve[i];
}
static void clip_to_window(AudioPsyClipContext *s,
const float *windowed_frame, float *clipping_delta, float delta_boost)
{
const float *window = s->window;
for (int i = 0; i < s->fft_size; i++) {
const float limit = s->clip_level * window[i];
const float effective_value = windowed_frame[i] + clipping_delta[i];
if (effective_value > limit) {
clipping_delta[i] += (limit - effective_value) * delta_boost;
} else if (effective_value < -limit) {
clipping_delta[i] += (-limit - effective_value) * delta_boost;
}
}
}
static void limit_clip_spectrum(AudioPsyClipContext *s,
float *clip_spectrum, const float *mask_curve)
{
// bin 0
float relative_distortion_level = FFABS(clip_spectrum[0]) / mask_curve[0];
if (relative_distortion_level > 1.f)
clip_spectrum[0] /= relative_distortion_level;
// bin 1..N/2-1
for (int i = 1; i < s->fft_size / 2; i++) {
float real = clip_spectrum[i * 2];
float imag = clip_spectrum[i * 2 + 1];
// although the negative frequencies are omitted because they are redundant,
// the magnitude of the positive frequencies are not doubled.
// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
relative_distortion_level = hypotf(real, imag) * 2 / mask_curve[i];
if (relative_distortion_level > 1.0) {
clip_spectrum[i * 2] /= relative_distortion_level;
clip_spectrum[i * 2 + 1] /= relative_distortion_level;
}
}
// bin N/2
relative_distortion_level = FFABS(clip_spectrum[1]) / mask_curve[s->fft_size / 2];
if (relative_distortion_level > 1.f)
clip_spectrum[1] /= relative_distortion_level;
}
static void r2c(float *buffer, int size)
{
for (int i = size - 1; i >= 0; i--)
buffer[2 * i] = buffer[i];
for (int i = size - 1; i >= 0; i--)
buffer[2 * i + 1] = 0.f;
}
static void c2r(float *buffer, int size)
{
for (int i = 0; i < size; i++)
buffer[i] = buffer[2 * i];
for (int i = 0; i < size; i++)
buffer[i + size] = 0.f;
}
static void feed(AVFilterContext *ctx, int ch,
const float *in_samples, float *out_samples, int diff_only,
float *in_frame, float *out_dist_frame,
float *windowed_frame, float *clipping_delta,
float *spectrum_buf, float *mask_curve)
{
AudioPsyClipContext *s = ctx->priv;
const float clip_level_inv = 1.f / s->clip_level;
const float level_out = s->level_out;
float orig_peak = 0;
float peak;
// shift in/out buffers
for (int i = 0; i < s->fft_size - s->overlap; i++) {
in_frame[i] = in_frame[i + s->overlap];
out_dist_frame[i] = out_dist_frame[i + s->overlap];
}
for (int i = 0; i < s->overlap; i++) {
in_frame[i + s->fft_size - s->overlap] = in_samples[i];
out_dist_frame[i + s->fft_size - s->overlap] = 0.f;
}
apply_window(s, in_frame, windowed_frame, 0);
r2c(windowed_frame, s->fft_size);
s->tx_fn(s->tx_ctx[ch], spectrum_buf, windowed_frame, sizeof(float));
c2r(windowed_frame, s->fft_size);
calculate_mask_curve(s, spectrum_buf, mask_curve);
// It would be easier to calculate the peak from the unwindowed input.
// This is just for consistency with the clipped peak calculateion
// because the inv_window zeros out samples on the edge of the window.
for (int i = 0; i < s->fft_size; i++)
orig_peak = FFMAX(orig_peak, FFABS(windowed_frame[i] * s->inv_window[i]));
orig_peak *= clip_level_inv;
peak = orig_peak;
// clear clipping_delta
for (int i = 0; i < s->fft_size * 2; i++)
clipping_delta[i] = 0.f;
// repeat clipping-filtering process a few times to control both the peaks and the spectrum
for (int i = 0; i < s->iterations; i++) {
float mask_curve_shift = 1.122f; // 1.122 is 1dB
// The last 1/3 of rounds have boosted delta to help reach the peak target faster
float delta_boost = 1.f;
if (i >= s->iterations - s->iterations / 3) {
// boosting the delta when largs peaks are still present is dangerous
if (peak < 2.f)
delta_boost = 2.f;
}
clip_to_window(s, windowed_frame, clipping_delta, delta_boost);
r2c(clipping_delta, s->fft_size);
s->tx_fn(s->tx_ctx[ch], spectrum_buf, clipping_delta, sizeof(float));
limit_clip_spectrum(s, spectrum_buf, mask_curve);
s->itx_fn(s->itx_ctx[ch], clipping_delta, spectrum_buf, sizeof(float));
c2r(clipping_delta, s->fft_size);
for (int i = 0; i < s->fft_size; i++)
clipping_delta[i] /= s->fft_size;
peak = 0;
for (int i = 0; i < s->fft_size; i++)
peak = FFMAX(peak, FFABS((windowed_frame[i] + clipping_delta[i]) * s->inv_window[i]));
peak *= clip_level_inv;
// Automatically adjust mask_curve as necessary to reach peak target
if (orig_peak > 1.f && peak > 1.f) {
float diff_achieved = orig_peak - peak;
if (i + 1 < s->iterations - s->iterations / 3 && diff_achieved > 0) {
float diff_needed = orig_peak - 1.f;
float diff_ratio = diff_needed / diff_achieved;
// If a good amount of peak reduction was already achieved,
// don't shift the mask_curve by the full peak value
// On the other hand, if only a little peak reduction was achieved,
// don't shift the mask_curve by the enormous diff_ratio.
diff_ratio = FFMIN(diff_ratio, peak);
mask_curve_shift = FFMAX(mask_curve_shift, diff_ratio);
} else {
// If the peak got higher than the input or we are in the last 1/3 rounds,
// go back to the heavy-handed peak heuristic.
mask_curve_shift = FFMAX(mask_curve_shift, peak);
}
}
mask_curve_shift = 1.f + (mask_curve_shift - 1.f) * s->adaptive;
// Be less strict in the next iteration.
// This helps with peak control.
for (int i = 0; i < s->fft_size / 2 + 1; i++)
mask_curve[i] *= mask_curve_shift;
}
// do overlap & add
apply_window(s, clipping_delta, out_dist_frame, 1);
for (int i = 0; i < s->overlap; i++) {
// 4 times overlap with squared hanning window results in 1.5 time increase in amplitude
if (!ctx->is_disabled) {
out_samples[i] = out_dist_frame[i] / 1.5f;
if (!diff_only)
out_samples[i] += in_frame[i];
if (s->auto_level)
out_samples[i] *= clip_level_inv;
out_samples[i] *= level_out;
} else {
out_samples[i] = in_frame[i];
}
}
}
static int psy_channel(AVFilterContext *ctx, AVFrame *in, AVFrame *out, int ch)
{
AudioPsyClipContext *s = ctx->priv;
const float *src = (const float *)in->extended_data[ch];
float *in_buffer = (float *)s->in_buffer->extended_data[ch];
float *dst = (float *)out->extended_data[ch];
for (int n = 0; n < s->overlap; n++)
in_buffer[n] = src[n] * s->level_in;
feed(ctx, ch, in_buffer, dst, s->diff_only,
(float *)(s->in_frame->extended_data[ch]),
(float *)(s->out_dist_frame->extended_data[ch]),
(float *)(s->windowed_frame->extended_data[ch]),
(float *)(s->clipping_delta->extended_data[ch]),
(float *)(s->spectrum_buf->extended_data[ch]),
(float *)(s->mask_curve->extended_data[ch]));
return 0;
}
static int psy_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioPsyClipContext *s = ctx->priv;
AVFrame *out = arg;
const int start = (out->channels * jobnr) / nb_jobs;
const int end = (out->channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++)
psy_channel(ctx, s->in, out, ch);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioPsyClipContext *s = ctx->priv;
AVFrame *out;
int ret;
out = ff_get_audio_buffer(outlink, s->overlap);
if (!out) {
ret = AVERROR(ENOMEM);
goto fail;
}
s->in = in;
ff_filter_execute(ctx, psy_channels, out, NULL,
FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
out->pts = in->pts;
out->nb_samples = in->nb_samples;
ret = ff_filter_frame(outlink, out);
fail:
av_frame_free(&in);
s->in = NULL;
return ret < 0 ? ret : 0;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioPsyClipContext *s = ctx->priv;
AVFrame *in = NULL;
int ret = 0, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->overlap, s->overlap, &in);
if (ret < 0)
return ret;
if (ret > 0) {
return filter_frame(inlink, in);
} else if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
} else {
if (ff_inlink_queued_samples(inlink) >= s->overlap) {
ff_filter_set_ready(ctx, 10);
} else if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(inlink);
}
return 0;
}
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioPsyClipContext *s = ctx->priv;
av_freep(&s->window);
av_freep(&s->inv_window);
av_freep(&s->spread_table);
av_freep(&s->spread_table_range);
av_freep(&s->spread_table_index);
av_freep(&s->margin_curve);
av_frame_free(&s->in_buffer);
av_frame_free(&s->in_frame);
av_frame_free(&s->out_dist_frame);
av_frame_free(&s->windowed_frame);
av_frame_free(&s->clipping_delta);
av_frame_free(&s->spectrum_buf);
av_frame_free(&s->mask_curve);
for (int ch = 0; ch < s->channels; ch++) {
if (s->tx_ctx)
av_tx_uninit(&s->tx_ctx[ch]);
if (s->itx_ctx)
av_tx_uninit(&s->itx_ctx[ch]);
}
av_freep(&s->tx_ctx);
av_freep(&s->itx_ctx);
}
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_apsyclip = {
.name = "apsyclip",
.description = NULL_IF_CONFIG_SMALL("Audio Psychoacoustic Clipper."),
.query_formats = query_formats,
.priv_size = sizeof(AudioPsyClipContext),
.priv_class = &apsyclip_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.activate = activate,
.process_command = ff_filter_process_command,
};

1
libavfilter/allfilters.c

@ -67,6 +67,7 @@ extern const AVFilter ff_af_apad;
extern const AVFilter ff_af_aperms;
extern const AVFilter ff_af_aphaser;
extern const AVFilter ff_af_aphaseshift;
extern const AVFilter ff_af_apsyclip;
extern const AVFilter ff_af_apulsator;
extern const AVFilter ff_af_arealtime;
extern const AVFilter ff_af_aresample;

2
libavfilter/version.h

@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
#define LIBAVFILTER_VERSION_MINOR 8
#define LIBAVFILTER_VERSION_MINOR 9
#define LIBAVFILTER_VERSION_MICRO 100

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