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b4f5201967
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>
241 lines
8.1 KiB
C
241 lines
8.1 KiB
C
/*
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* Copyright (c) 2011 Mina Nagy Zaki
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* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
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* This source code is freely redistributable and may be used for any purpose.
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* This copyright notice must be maintained. Edward Beingessner And Sundry
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* Contributors are not responsible for the consequences of using this
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* software.
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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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/**
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* @file
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* Stereo Widening Effect. Adds audio cues to move stereo image in
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* front of the listener. Adapted from the libsox earwax effect.
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*/
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#include "libavutil/channel_layout.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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#define NUMTAPS 32
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static const int8_t filt[NUMTAPS * 2] = {
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/* 30° 330° */
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4, -6, /* 32 tap stereo FIR filter. */
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4, -11, /* One side filters as if the */
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-1, -5, /* signal was from 30 degrees */
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3, 3, /* from the ear, the other as */
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-2, 5, /* if 330 degrees. */
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-5, 0,
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9, 1,
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6, 3, /* Input */
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-4, -1, /* Left Right */
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-5, -3, /* __________ __________ */
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-2, -5, /* | | | | */
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-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
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6, -7, /* / |__________| |__________| \ */
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30, -29, /* / \ / \ */
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12, -3, /* / X \ */
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-11, 4, /* / / \ \ */
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-3, 7, /* ____V_____ __________V V__________ _____V____ */
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-20, 23, /* | | | | | | | | */
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2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
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1, -6, /* |__________| |__________| |__________| |__________| */
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-14, -5, /* \ ___ / \ ___ / */
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15, -18, /* \ / \ / _____ \ / \ / */
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6, 7, /* `->| + |<--' / \ `-->| + |<-' */
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15, -10, /* \___/ _/ \_ \___/ */
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-14, 22, /* \ / \ / \ / */
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-7, -2, /* `--->| | | |<---' */
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-4, 9, /* \_/ \_/ */
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6, -12, /* */
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6, -6, /* Headphones */
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0, -11,
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0, -5,
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4, 0};
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typedef struct EarwaxContext {
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int16_t filter[2][NUMTAPS];
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int16_t taps[4][NUMTAPS * 2];
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AVFrame *frame[2];
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} EarwaxContext;
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static int query_formats(AVFilterContext *ctx)
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{
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static const int sample_rates[] = { 44100, -1 };
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int ret;
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
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(ret = ff_set_common_samplerates_from_list(ctx, sample_rates)) < 0)
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return ret;
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return 0;
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}
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//FIXME: replace with DSPContext.scalarproduct_int16
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static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
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const int16_t *filt, int16_t *out)
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{
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int32_t sample;
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int16_t j;
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while (in < endin) {
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sample = 0;
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for (j = 0; j < NUMTAPS; j++)
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sample += in[j] * filt[j];
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*out = av_clip_int16(sample >> 7);
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out++;
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in++;
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}
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return out;
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}
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static int config_input(AVFilterLink *inlink)
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{
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EarwaxContext *s = inlink->dst->priv;
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for (int i = 0; i < NUMTAPS; i++) {
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s->filter[0][i] = filt[i * 2];
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s->filter[1][i] = filt[i * 2 + 1];
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}
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return 0;
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}
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static void convolve(AVFilterContext *ctx, AVFrame *in,
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int input_ch, int output_ch,
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int filter_ch, int tap_ch)
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{
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EarwaxContext *s = ctx->priv;
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int16_t *taps, *endin, *dst, *src;
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int len;
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taps = s->taps[tap_ch];
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dst = (int16_t *)s->frame[input_ch]->data[output_ch];
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src = (int16_t *)in->data[input_ch];
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len = FFMIN(NUMTAPS, in->nb_samples);
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// copy part of new input and process with saved input
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memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
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dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
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// process current input
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if (in->nb_samples >= NUMTAPS) {
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endin = src + in->nb_samples - NUMTAPS;
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scalarproduct(src, endin, s->filter[filter_ch], dst);
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// save part of input for next round
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memcpy(taps, endin, NUMTAPS * sizeof(*taps));
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} else {
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memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
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}
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}
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static void mix(AVFilterContext *ctx, AVFrame *out,
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int output_ch, int f0, int f1, int i0, int i1)
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{
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EarwaxContext *s = ctx->priv;
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const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
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const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
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int16_t *dst = (int16_t *)out->data[output_ch];
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for (int n = 0; n < out->nb_samples; n++)
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dst[n] = av_clip_int16(srcl[n] + srcr[n]);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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EarwaxContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
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for (int ch = 0; ch < 2; ch++) {
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if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
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av_frame_free(&s->frame[ch]);
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s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!s->frame[ch]) {
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av_frame_free(&in);
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av_frame_free(&out);
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return AVERROR(ENOMEM);
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}
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}
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}
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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convolve(ctx, in, 0, 0, 0, 0);
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convolve(ctx, in, 0, 1, 1, 1);
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convolve(ctx, in, 1, 0, 0, 2);
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convolve(ctx, in, 1, 1, 1, 3);
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mix(ctx, out, 0, 0, 1, 1, 0);
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mix(ctx, out, 1, 0, 1, 0, 1);
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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EarwaxContext *s = ctx->priv;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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}
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static const AVFilterPad earwax_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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};
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static const AVFilterPad earwax_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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};
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const AVFilter ff_af_earwax = {
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.name = "earwax",
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.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
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.priv_size = sizeof(EarwaxContext),
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.uninit = uninit,
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FILTER_INPUTS(earwax_inputs),
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FILTER_OUTPUTS(earwax_outputs),
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FILTER_QUERY_FUNC(query_formats),
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};
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