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avfilter: add audio psychoacoustic clipper
This commit is contained in:
Paul B Mahol
parent
966fc3c070
commit
eeab62ad2d
@ -19,6 +19,7 @@ version <next>:
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- swscale slice threading
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- MSN Siren decoder
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- scharr video filter
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- apsyclip audio filter
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version 4.4:
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@ -2363,6 +2363,42 @@ Default value is 8.
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This filter supports the all above options as @ref{commands}.
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@section apsyclip
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Apply Psychoacoustic clipper to input audio stream.
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The filter accepts the following options:
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@table @option
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@item level_in
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Set input gain. By default it is 1. Range is [0.015625 - 64].
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@item level_out
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Set output gain. By default it is 1. Range is [0.015625 - 64].
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@item clip
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Set the clipping start value. Default value is 0dBFS or 1.
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@item diff
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Output only difference samples, useful to hear introduced distortions.
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By default is disabled.
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@item adaptive
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Set strenght of adaptive distortion applied. Default value is 0.5.
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Allowed range is from 0 to 1.
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@item iterations
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Set number of iterations of psychoacoustic clipper.
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Allowed range is from 1 to 20. Default value is 10.
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@item level
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Auto level output signal. Default is disabled.
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This normalizes audio back to 0dBFS if enabled.
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@end table
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@subsection Commands
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This filter supports the all above options as @ref{commands}.
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@section apulsator
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Audio pulsator is something between an autopanner and a tremolo.
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@ -74,6 +74,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
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OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
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OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
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OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
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OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
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OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
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OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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679
libavfilter/af_apsyclip.c
Normal file
679
libavfilter/af_apsyclip.c
Normal file
@ -0,0 +1,679 @@
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/*
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* Copyright (c) 2014 - 2021 Jason Jang
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* Copyright (c) 2021 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public License
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* as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "libavutil/tx.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "internal.h"
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typedef struct AudioPsyClipContext {
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const AVClass *class;
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double level_in;
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double level_out;
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double clip_level;
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double adaptive;
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int auto_level;
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int diff_only;
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int iterations;
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char *protections_str;
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double *protections;
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int num_psy_bins;
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int fft_size;
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int overlap;
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int channels;
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int spread_table_rows;
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int *spread_table_index;
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int (*spread_table_range)[2];
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float *window, *inv_window, *spread_table, *margin_curve;
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AVFrame *in;
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AVFrame *in_buffer;
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AVFrame *in_frame;
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AVFrame *out_dist_frame;
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AVFrame *windowed_frame;
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AVFrame *clipping_delta;
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AVFrame *spectrum_buf;
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AVFrame *mask_curve;
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AVTXContext **tx_ctx;
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av_tx_fn tx_fn;
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AVTXContext **itx_ctx;
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av_tx_fn itx_fn;
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} AudioPsyClipContext;
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#define OFFSET(x) offsetof(AudioPsyClipContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption apsyclip_options[] = {
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{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
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{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, FLAGS },
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{ "clip", "set clip level", OFFSET(clip_level), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 1, FLAGS },
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{ "diff", "enable difference", OFFSET(diff_only), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
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{ "adaptive", "set adaptive distortion", OFFSET(adaptive), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, FLAGS },
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{ "iterations", "set iterations", OFFSET(iterations), AV_OPT_TYPE_INT, {.i64=10}, 1, 20, FLAGS },
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{ "level", "set auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
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{NULL}
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};
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AVFILTER_DEFINE_CLASS(apsyclip);
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static int query_formats(AVFilterContext *ctx)
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{
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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ret = ff_set_common_all_channel_counts(ctx);
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if (ret < 0)
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return ret;
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ret = ff_set_common_formats_from_list(ctx, sample_fmts);
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if (ret < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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}
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static void generate_hann_window(float *window, float *inv_window, int size)
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{
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for (int i = 0; i < size; i++) {
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float value = 0.5f * (1.f - cosf(2.f * M_PI * i / size));
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window[i] = value;
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// 1/window to calculate unwindowed peak.
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inv_window[i] = value > 0.01f ? 1.f / value : 0.f;
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}
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}
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static void set_margin_curve(AudioPsyClipContext *s,
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const int (*points)[2], int num_points, int sample_rate)
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{
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int j = 0;
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s->margin_curve[0] = points[0][1];
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for (int i = 0; i < num_points - 1; i++) {
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while (j < s->fft_size / 2 + 1 && j * sample_rate / s->fft_size < points[i + 1][0]) {
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// linearly interpolate between points
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int binHz = j * sample_rate / s->fft_size;
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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]);
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j++;
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}
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}
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// handle bins after the last point
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while (j < s->fft_size / 2 + 1) {
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s->margin_curve[j] = points[num_points - 1][1];
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j++;
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}
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// convert margin curve to linear amplitude scale
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for (j = 0; j < s->fft_size / 2 + 1; j++)
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s->margin_curve[j] = powf(10.f, s->margin_curve[j] / 20.f);
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}
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static void generate_spread_table(AudioPsyClipContext *s)
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{
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// Calculate tent-shape function in log-log scale.
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// As an optimization, only consider bins close to "bin"
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// This reduces the number of multiplications needed in calculate_mask_curve
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// The masking contribution at faraway bins is negligeable
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// Another optimization to save memory and speed up the calculation of the
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// spread table is to calculate and store only 2 spread functions per
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// octave, and reuse the same spread function for multiple bins.
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int table_index = 0;
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int bin = 0;
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int increment = 1;
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while (bin < s->num_psy_bins) {
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float sum = 0;
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int base_idx = table_index * s->num_psy_bins;
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int start_bin = bin * 3 / 4;
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int end_bin = FFMIN(s->num_psy_bins, ((bin + 1) * 4 + 2) / 3);
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int next_bin;
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for (int j = start_bin; j < end_bin; j++) {
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// add 0.5 so i=0 doesn't get log(0)
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float rel_idx_log = FFABS(logf((j + 0.5f) / (bin + 0.5f)));
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float value;
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if (j >= bin) {
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// mask up
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value = expf(-rel_idx_log * 40.f);
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} else {
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// mask down
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value = expf(-rel_idx_log * 80.f);
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}
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// the spreading function is centred in the row
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sum += value;
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s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] = value;
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}
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// now normalize it
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for (int j = start_bin; j < end_bin; j++) {
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s->spread_table[base_idx + s->num_psy_bins / 2 + j - bin] /= sum;
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}
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s->spread_table_range[table_index][0] = start_bin - bin;
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s->spread_table_range[table_index][1] = end_bin - bin;
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if (bin <= 1) {
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next_bin = bin + 1;
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} else {
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if ((bin & (bin - 1)) == 0) {
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// power of 2
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increment = bin / 2;
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}
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next_bin = bin + increment;
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}
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// set bins between "bin" and "next_bin" to use this table_index
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for (int i = bin; i < next_bin; i++)
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s->spread_table_index[i] = table_index;
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bin = next_bin;
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table_index++;
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}
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioPsyClipContext *s = ctx->priv;
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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} };
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static const int num_points = 10;
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float scale;
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int ret;
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s->fft_size = inlink->sample_rate > 100000 ? 1024 : inlink->sample_rate > 50000 ? 512 : 256;
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s->overlap = s->fft_size / 4;
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// The psy masking calculation is O(n^2),
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// so skip it for frequencies not covered by base sampling rantes (i.e. 44k)
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if (inlink->sample_rate <= 50000) {
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s->num_psy_bins = s->fft_size / 2;
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} else if (inlink->sample_rate <= 100000) {
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s->num_psy_bins = s->fft_size / 4;
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} else {
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s->num_psy_bins = s->fft_size / 8;
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}
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s->window = av_calloc(s->fft_size, sizeof(*s->window));
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s->inv_window = av_calloc(s->fft_size, sizeof(*s->inv_window));
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if (!s->window || !s->inv_window)
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return AVERROR(ENOMEM);
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s->in_buffer = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->in_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->out_dist_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->windowed_frame = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->clipping_delta = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->spectrum_buf = ff_get_audio_buffer(inlink, s->fft_size * 2);
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s->mask_curve = ff_get_audio_buffer(inlink, s->fft_size / 2 + 1);
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if (!s->in_buffer || !s->in_frame ||
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!s->out_dist_frame || !s->windowed_frame ||
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!s->clipping_delta || !s->spectrum_buf || !s->mask_curve)
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return AVERROR(ENOMEM);
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generate_hann_window(s->window, s->inv_window, s->fft_size);
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s->margin_curve = av_calloc(s->fft_size / 2 + 1, sizeof(*s->margin_curve));
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if (!s->margin_curve)
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return AVERROR(ENOMEM);
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s->spread_table_rows = av_log2(s->num_psy_bins) * 2;
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s->spread_table = av_calloc(s->spread_table_rows * s->num_psy_bins, sizeof(*s->spread_table));
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if (!s->spread_table)
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return AVERROR(ENOMEM);
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s->spread_table_range = av_calloc(s->spread_table_rows * 2, sizeof(*s->spread_table_range));
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if (!s->spread_table_range)
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return AVERROR(ENOMEM);
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s->spread_table_index = av_calloc(s->num_psy_bins, sizeof(*s->spread_table_index));
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if (!s->spread_table_index)
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return AVERROR(ENOMEM);
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set_margin_curve(s, points, num_points, inlink->sample_rate);
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generate_spread_table(s);
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s->channels = inlink->channels;
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s->tx_ctx = av_calloc(s->channels, sizeof(*s->tx_ctx));
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s->itx_ctx = av_calloc(s->channels, sizeof(*s->itx_ctx));
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if (!s->tx_ctx || !s->itx_ctx)
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return AVERROR(ENOMEM);
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for (int ch = 0; ch < s->channels; ch++) {
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ret = av_tx_init(&s->tx_ctx[ch], &s->tx_fn, AV_TX_FLOAT_FFT, 0, s->fft_size, &scale, 0);
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if (ret < 0)
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return ret;
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ret = av_tx_init(&s->itx_ctx[ch], &s->itx_fn, AV_TX_FLOAT_FFT, 1, s->fft_size, &scale, 0);
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if (ret < 0)
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return ret;
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}
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return 0;
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}
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static void apply_window(AudioPsyClipContext *s,
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const float *in_frame, float *out_frame, const int add_to_out_frame)
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{
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const float *window = s->window;
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for (int i = 0; i < s->fft_size; i++) {
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if (add_to_out_frame) {
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out_frame[i] += in_frame[i] * window[i];
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} else {
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out_frame[i] = in_frame[i] * window[i];
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}
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}
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}
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static void calculate_mask_curve(AudioPsyClipContext *s,
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const float *spectrum, float *mask_curve)
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{
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for (int i = 0; i < s->fft_size / 2 + 1; i++)
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mask_curve[i] = 0;
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for (int i = 0; i < s->num_psy_bins; i++) {
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float magnitude;
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int table_idx;
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int range[2];
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if (i == 0) {
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magnitude = FFABS(spectrum[0]);
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} else if (i == s->fft_size / 2) {
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magnitude = FFABS(spectrum[1]);
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} else {
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// although the negative frequencies are omitted because they are redundant,
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// the magnitude of the positive frequencies are not doubled.
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// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
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magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
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}
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table_idx = s->spread_table_index[i];
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range[0] = s->spread_table_range[table_idx][0];
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range[1] = s->spread_table_range[table_idx][1];
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int base_idx = table_idx * s->num_psy_bins;
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int start_bin = FFMAX(0, i + range[0]);
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int end_bin = FFMIN(s->num_psy_bins, i + range[1]);
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for (int j = start_bin; j < end_bin; j++)
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mask_curve[j] += s->spread_table[base_idx + s->num_psy_bins / 2 + j - i] * magnitude;
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}
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// for ultrasonic frequencies, skip the O(n^2) spread calculation and just copy the magnitude
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for (int i = s->num_psy_bins; i < s->fft_size / 2 + 1; i++) {
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float magnitude;
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if (i == s->fft_size / 2) {
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magnitude = FFABS(spectrum[1]);
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} else {
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// although the negative frequencies are omitted because they are redundant,
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// the magnitude of the positive frequencies are not doubled.
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// Multiply the magnitude by 2 to simulate adding up the + and - frequencies.
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magnitude = hypotf(spectrum[2 * i], spectrum[2 * i + 1]) * 2;
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}
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mask_curve[i] = magnitude;
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}
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for (int i = 0; i < s->fft_size / 2 + 1; i++)
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mask_curve[i] = mask_curve[i] / s->margin_curve[i];
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}
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static void clip_to_window(AudioPsyClipContext *s,
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const float *windowed_frame, float *clipping_delta, float delta_boost)
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{
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const float *window = s->window;
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for (int i = 0; i < s->fft_size; i++) {
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const float limit = s->clip_level * window[i];
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const float effective_value = windowed_frame[i] + clipping_delta[i];
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if (effective_value > limit) {
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clipping_delta[i] += (limit - effective_value) * delta_boost;
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} else if (effective_value < -limit) {
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clipping_delta[i] += (-limit - effective_value) * delta_boost;
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}
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}
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}
|
||||
|
||||
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,
|
||||
};
|
||||
@ -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;
|
||||
|
||||
@ -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
|
||||
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user