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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
209 lines
7.8 KiB
C
209 lines
7.8 KiB
C
/*
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* Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
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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
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* License 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 GNU
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* 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
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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#include "generate_wave_table.h"
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#define INTERPOLATION_LINEAR 0
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#define INTERPOLATION_QUADRATIC 1
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typedef struct FlangerContext {
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const AVClass *class;
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double delay_min;
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double delay_depth;
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double feedback_gain;
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double delay_gain;
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double speed;
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int wave_shape;
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double channel_phase;
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int interpolation;
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double in_gain;
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int max_samples;
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uint8_t **delay_buffer;
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int delay_buf_pos;
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double *delay_last;
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float *lfo;
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int lfo_length;
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int lfo_pos;
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} FlangerContext;
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#define OFFSET(x) offsetof(FlangerContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption flanger_options[] = {
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{ "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
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{ "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
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{ "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
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{ "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
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{ "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
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{ "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, .unit = "type" },
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{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" },
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{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, .unit = "type" },
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{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "type" },
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{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, .unit = "type" },
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{ "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
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{ "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, .unit = "itype" },
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{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, .unit = "itype" },
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{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, .unit = "itype" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(flanger);
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static av_cold int init(AVFilterContext *ctx)
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{
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FlangerContext *s = ctx->priv;
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s->feedback_gain /= 100;
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s->delay_gain /= 100;
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s->channel_phase /= 100;
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s->delay_min /= 1000;
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s->delay_depth /= 1000;
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s->in_gain = 1 / (1 + s->delay_gain);
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s->delay_gain /= 1 + s->delay_gain;
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s->delay_gain *= 1 - fabs(s->feedback_gain);
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return 0;
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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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FlangerContext *s = ctx->priv;
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s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
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s->lfo_length = inlink->sample_rate / s->speed;
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s->delay_last = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->delay_last));
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s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
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if (!s->lfo || !s->delay_last)
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return AVERROR(ENOMEM);
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ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
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rint(s->delay_min * inlink->sample_rate),
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s->max_samples - 2., 3 * M_PI_2);
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return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
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inlink->ch_layout.nb_channels, s->max_samples,
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inlink->format, 0);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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FlangerContext *s = ctx->priv;
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AVFrame *out_frame;
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int chan, i;
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if (av_frame_is_writable(frame)) {
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out_frame = frame;
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} else {
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out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
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if (!out_frame) {
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av_frame_free(&frame);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out_frame, frame);
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}
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for (i = 0; i < frame->nb_samples; i++) {
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s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
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for (chan = 0; chan < inlink->ch_layout.nb_channels; chan++) {
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double *src = (double *)frame->extended_data[chan];
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double *dst = (double *)out_frame->extended_data[chan];
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double delayed_0, delayed_1;
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double delayed;
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double in, out;
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int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
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double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
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int int_delay = (int)delay;
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double frac_delay = modf(delay, &delay);
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double *delay_buffer = (double *)s->delay_buffer[chan];
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in = src[i];
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delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
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s->feedback_gain;
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delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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if (s->interpolation == INTERPOLATION_LINEAR) {
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delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
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} else {
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double a, b;
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double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
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delayed_2 -= delayed_0;
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delayed_1 -= delayed_0;
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a = delayed_2 * .5 - delayed_1;
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b = delayed_1 * 2 - delayed_2 *.5;
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delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
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}
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s->delay_last[chan] = delayed;
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out = in * s->in_gain + delayed * s->delay_gain;
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dst[i] = out;
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}
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s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
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}
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if (frame != out_frame)
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av_frame_free(&frame);
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return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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FlangerContext *s = ctx->priv;
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av_freep(&s->lfo);
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av_freep(&s->delay_last);
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if (s->delay_buffer)
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av_freep(&s->delay_buffer[0]);
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av_freep(&s->delay_buffer);
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}
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static const AVFilterPad flanger_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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.filter_frame = filter_frame,
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},
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};
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const AVFilter ff_af_flanger = {
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.name = "flanger",
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.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
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.priv_size = sizeof(FlangerContext),
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.priv_class = &flanger_class,
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.init = init,
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.uninit = uninit,
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FILTER_INPUTS(flanger_inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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};
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