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ad60b3b181
* qatar/master: vorbis: Validate that the floor 1 X values contain no duplicates. avprobe: Identify codec probe failures rather than calling them unsupported codecs. avformat: Probe codecs at score 0 on buffer exhaustion conditions. avformat: Factorize codec probing. Indeo Audio decoder imc: make IMDCT support stereo output imc: move channel-specific data into separate context lavfi: remove request/poll and drawing functions from public API on next bump lavfi: make avfilter_insert_pad and pals private on next bump. lavfi: make formats API private on next bump. avplay: use buffersrc instead of custom input filter. avtools: move buffer management code from avconv to cmdutils. avconv: don't use InputStream in the buffer management code. avconv: fix exiting when max frames is reached. mpc8: fix maximum bands handling aacdec: Turn PS off when switching to stereo and turn it to implicit when switching to mono. Conflicts: Changelog cmdutils.h ffmpeg.c ffplay.c ffprobe.c libavcodec/avcodec.h libavcodec/mpc8.c libavcodec/v210dec.h libavcodec/version.h libavcodec/vorbisdec.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersrc.c libavfilter/formats.c libavfilter/src_movie.c libavfilter/vf_aspect.c libavfilter/vf_blackframe.c libavfilter/vf_boxblur.c libavfilter/vf_crop.c libavfilter/vf_cropdetect.c libavfilter/vf_delogo.c libavfilter/vf_drawbox.c libavfilter/vf_drawtext.c libavfilter/vf_fade.c libavfilter/vf_fifo.c libavfilter/vf_format.c libavfilter/vf_frei0r.c libavfilter/vf_gradfun.c libavfilter/vf_hflip.c libavfilter/vf_hqdn3d.c libavfilter/vf_libopencv.c libavfilter/vf_lut.c libavfilter/vf_overlay.c libavfilter/vf_pad.c libavfilter/vf_scale.c libavfilter/vf_select.c libavfilter/vf_showinfo.c libavfilter/vf_transpose.c libavfilter/vf_unsharp.c libavfilter/vf_yadif.c libavfilter/vsrc_color.c libavfilter/vsrc_testsrc.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
210 lines
6.5 KiB
C
210 lines
6.5 KiB
C
/*
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* Copyright (c) 2011 Stefano Sabatini
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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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* buffer sink
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*/
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#include "libavutil/audio_fifo.h"
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#include "libavutil/audioconvert.h"
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#include "libavutil/fifo.h"
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#include "libavutil/mathematics.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "buffersink.h"
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#include "internal.h"
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typedef struct {
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AVFifoBuffer *fifo; ///< FIFO buffer of frame references
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AVAudioFifo *audio_fifo; ///< FIFO for audio samples
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int64_t next_pts; ///< interpolating audio pts
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} BufferSinkContext;
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#define FIFO_INIT_SIZE 8
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static av_cold void uninit(AVFilterContext *ctx)
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{
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BufferSinkContext *sink = ctx->priv;
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while (sink->fifo && av_fifo_size(sink->fifo)) {
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AVFilterBufferRef *buf;
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av_fifo_generic_read(sink->fifo, &buf, sizeof(buf), NULL);
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avfilter_unref_buffer(buf);
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}
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av_fifo_free(sink->fifo);
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if (sink->audio_fifo)
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av_audio_fifo_free(sink->audio_fifo);
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}
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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{
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BufferSinkContext *sink = ctx->priv;
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if (!(sink->fifo = av_fifo_alloc(FIFO_INIT_SIZE*sizeof(AVFilterBufferRef*)))) {
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av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo\n");
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return AVERROR(ENOMEM);
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}
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return 0;
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}
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static void write_buf(AVFilterContext *ctx, AVFilterBufferRef *buf)
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{
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BufferSinkContext *sink = ctx->priv;
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if (av_fifo_space(sink->fifo) < sizeof(AVFilterBufferRef *) &&
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(av_fifo_realloc2(sink->fifo, av_fifo_size(sink->fifo) * 2) < 0)) {
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av_log(ctx, AV_LOG_ERROR, "Error reallocating the FIFO.\n");
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return;
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}
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av_fifo_generic_write(sink->fifo, &buf, sizeof(buf), NULL);
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}
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static void end_frame(AVFilterLink *link)
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{
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write_buf(link->dst, link->cur_buf);
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link->cur_buf = NULL;
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}
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static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
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{
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write_buf(link->dst, buf);
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}
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int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
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{
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BufferSinkContext *sink = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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int ret;
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if (!buf) {
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if (av_fifo_size(sink->fifo))
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return av_fifo_size(sink->fifo)/sizeof(*buf);
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else
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return ff_poll_frame(ctx->inputs[0]);
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}
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if (!av_fifo_size(sink->fifo) &&
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(ret = ff_request_frame(link)) < 0)
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return ret;
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if (!av_fifo_size(sink->fifo))
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return AVERROR(EINVAL);
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av_fifo_generic_read(sink->fifo, buf, sizeof(*buf), NULL);
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return 0;
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}
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static int read_from_fifo(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
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int nb_samples)
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{
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BufferSinkContext *s = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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AVFilterBufferRef *buf;
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if (!(buf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples)))
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return AVERROR(ENOMEM);
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av_audio_fifo_read(s->audio_fifo, (void**)buf->extended_data, nb_samples);
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buf->pts = s->next_pts;
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},
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link->time_base);
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*pbuf = buf;
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return 0;
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}
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int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
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int nb_samples)
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{
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BufferSinkContext *s = ctx->priv;
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AVFilterLink *link = ctx->inputs[0];
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int ret = 0;
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if (!s->audio_fifo) {
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int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
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if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))
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return AVERROR(ENOMEM);
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}
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while (ret >= 0) {
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AVFilterBufferRef *buf;
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if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)
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return read_from_fifo(ctx, pbuf, nb_samples);
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ret = av_buffersink_read(ctx, &buf);
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if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo))
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return read_from_fifo(ctx, pbuf, av_audio_fifo_size(s->audio_fifo));
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else if (ret < 0)
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return ret;
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if (buf->pts != AV_NOPTS_VALUE) {
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s->next_pts = buf->pts -
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av_rescale_q(av_audio_fifo_size(s->audio_fifo),
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(AVRational){ 1, link->sample_rate },
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link->time_base);
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}
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ret = av_audio_fifo_write(s->audio_fifo, (void**)buf->extended_data,
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buf->audio->nb_samples);
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avfilter_unref_buffer(buf);
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}
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return ret;
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}
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AVFilter avfilter_vsink_buffer = {
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.name = "buffersink_old",
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.description = NULL_IF_CONFIG_SMALL("Buffer video frames, and make them available to the end of the filter graph."),
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.priv_size = sizeof(BufferSinkContext),
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.init = init,
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.uninit = uninit,
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_VIDEO,
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.end_frame = end_frame,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL }},
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.outputs = (AVFilterPad[]) {{ .name = NULL }},
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};
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AVFilter avfilter_asink_abuffer = {
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.name = "abuffersink_old",
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.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them available to the end of the filter graph."),
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.priv_size = sizeof(BufferSinkContext),
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.init = init,
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.uninit = uninit,
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL }},
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.outputs = (AVFilterPad[]) {{ .name = NULL }},
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};
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