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7ceb9e6b11
Since the default in the libav fork is to only allow known layouts, making unknown layouts allowed by default here can be a security risk for filters directly merged from libav. However, usually it is simple to detect such cases, use of av_get_channel_layout_nb_channels is a good indicator, so I suggest we change this regardless. See http://ffmpeg.org/pipermail/ffmpeg-devel/2016-November/203204.html. This patch indirectly adds unknown channel layout support for filters where query_formats is not specified: abench afifo ainterleave anullsink apad aperms arealtime aselect asendcmd asetnsamples asetpts asettb ashowinfo azmq It introduces a query_formats callback for the asyncts filter, which only supports known channel layouts since it is using libavresample. And it removes .query_formats callback from filters where it was only there to support unknown layouts, as this is now the default: aloop ametadata anull asidedata asplit atrim Acked-by: Nicolas George <george@nsup.org> Signed-off-by: Marton Balint <cus@passwd.hu>
383 lines
10 KiB
C
383 lines
10 KiB
C
/*
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* Copyright (c) 2016 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
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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/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/fifo.h"
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#include "libavutil/internal.h"
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#include "libavutil/opt.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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#include "internal.h"
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#include "video.h"
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typedef struct LoopContext {
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const AVClass *class;
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AVAudioFifo *fifo;
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AVAudioFifo *left;
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AVFrame **frames;
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int nb_frames;
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int current_frame;
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int64_t start_pts;
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int64_t duration;
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int64_t current_sample;
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int64_t nb_samples;
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int64_t ignored_samples;
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int loop;
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int64_t size;
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int64_t start;
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int64_t pts;
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} LoopContext;
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#define AFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define VFLAGS AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OFFSET(x) offsetof(LoopContext, x)
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#if CONFIG_ALOOP_FILTER
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static int aconfig_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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LoopContext *s = ctx->priv;
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s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, 8192);
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s->left = av_audio_fifo_alloc(inlink->format, inlink->channels, 8192);
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if (!s->fifo || !s->left)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold void auninit(AVFilterContext *ctx)
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{
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LoopContext *s = ctx->priv;
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av_audio_fifo_free(s->fifo);
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av_audio_fifo_free(s->left);
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}
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static int push_samples(AVFilterContext *ctx, int nb_samples)
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{
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AVFilterLink *outlink = ctx->outputs[0];
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LoopContext *s = ctx->priv;
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AVFrame *out;
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int ret, i = 0;
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while (s->loop != 0 && i < nb_samples) {
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out = ff_get_audio_buffer(outlink, FFMIN(nb_samples, s->nb_samples - s->current_sample));
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if (!out)
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return AVERROR(ENOMEM);
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ret = av_audio_fifo_peek_at(s->fifo, (void **)out->extended_data, out->nb_samples, s->current_sample);
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if (ret < 0) {
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av_frame_free(&out);
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return ret;
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}
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out->pts = s->pts;
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out->nb_samples = ret;
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s->pts += out->nb_samples;
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i += out->nb_samples;
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s->current_sample += out->nb_samples;
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ret = ff_filter_frame(outlink, out);
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if (ret < 0)
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return ret;
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if (s->current_sample >= s->nb_samples) {
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s->current_sample = 0;
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if (s->loop > 0)
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s->loop--;
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}
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}
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return ret;
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}
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static int afilter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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LoopContext *s = ctx->priv;
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int ret = 0;
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if (s->ignored_samples + frame->nb_samples > s->start && s->size > 0 && s->loop != 0) {
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if (s->nb_samples < s->size) {
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int written = FFMIN(frame->nb_samples, s->size - s->nb_samples);
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int drain = 0;
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ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, written);
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if (ret < 0)
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return ret;
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if (!s->nb_samples) {
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drain = FFMAX(0, s->start - s->ignored_samples);
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s->pts = frame->pts;
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av_audio_fifo_drain(s->fifo, drain);
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s->pts += s->start - s->ignored_samples;
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}
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s->nb_samples += ret - drain;
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drain = frame->nb_samples - written;
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if (s->nb_samples == s->size && drain > 0) {
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int ret2;
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ret2 = av_audio_fifo_write(s->left, (void **)frame->extended_data, frame->nb_samples);
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if (ret2 < 0)
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return ret2;
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av_audio_fifo_drain(s->left, drain);
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}
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frame->nb_samples = ret;
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s->pts += ret;
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ret = ff_filter_frame(outlink, frame);
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} else {
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int nb_samples = frame->nb_samples;
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av_frame_free(&frame);
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ret = push_samples(ctx, nb_samples);
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}
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} else {
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s->ignored_samples += frame->nb_samples;
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frame->pts = s->pts;
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s->pts += frame->nb_samples;
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ret = ff_filter_frame(outlink, frame);
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}
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return ret;
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}
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static int arequest_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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LoopContext *s = ctx->priv;
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int ret = 0;
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if ((!s->size) ||
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(s->nb_samples < s->size) ||
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(s->nb_samples >= s->size && s->loop == 0)) {
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int nb_samples = av_audio_fifo_size(s->left);
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if (s->loop == 0 && nb_samples > 0) {
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AVFrame *out;
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out = ff_get_audio_buffer(outlink, nb_samples);
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if (!out)
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return AVERROR(ENOMEM);
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av_audio_fifo_read(s->left, (void **)out->extended_data, nb_samples);
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out->pts = s->pts;
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s->pts += nb_samples;
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ret = ff_filter_frame(outlink, out);
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if (ret < 0)
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return ret;
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}
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ret = ff_request_frame(ctx->inputs[0]);
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} else {
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ret = push_samples(ctx, 1024);
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}
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if (ret == AVERROR_EOF && s->nb_samples > 0 && s->loop != 0) {
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ret = push_samples(ctx, outlink->sample_rate);
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}
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return ret;
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}
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static const AVOption aloop_options[] = {
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{ "loop", "number of loops", OFFSET(loop), AV_OPT_TYPE_INT, {.i64 = 0 }, -1, INT_MAX, AFLAGS },
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{ "size", "max number of samples to loop", OFFSET(size), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT32_MAX, AFLAGS },
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{ "start", "set the loop start sample", OFFSET(start), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, AFLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aloop);
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static const AVFilterPad ainputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = afilter_frame,
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.config_props = aconfig_input,
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},
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{ NULL }
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};
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static const AVFilterPad aoutputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = arequest_frame,
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},
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{ NULL }
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};
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AVFilter ff_af_aloop = {
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.name = "aloop",
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.description = NULL_IF_CONFIG_SMALL("Loop audio samples."),
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.priv_size = sizeof(LoopContext),
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.priv_class = &aloop_class,
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.uninit = auninit,
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.inputs = ainputs,
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.outputs = aoutputs,
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};
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#endif /* CONFIG_ALOOP_FILTER */
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#if CONFIG_LOOP_FILTER
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static av_cold int init(AVFilterContext *ctx)
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{
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LoopContext *s = ctx->priv;
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s->frames = av_calloc(s->size, sizeof(*s->frames));
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if (!s->frames)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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LoopContext *s = ctx->priv;
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int i;
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for (i = 0; i < s->nb_frames; i++)
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av_frame_free(&s->frames[i]);
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av_freep(&s->frames);
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s->nb_frames = 0;
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}
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static int push_frame(AVFilterContext *ctx)
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{
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AVFilterLink *outlink = ctx->outputs[0];
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LoopContext *s = ctx->priv;
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int64_t pts;
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int ret;
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AVFrame *out = av_frame_clone(s->frames[s->current_frame]);
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if (!out)
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return AVERROR(ENOMEM);
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out->pts += s->duration - s->start_pts;
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pts = out->pts + av_frame_get_pkt_duration(out);
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ret = ff_filter_frame(outlink, out);
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s->current_frame++;
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if (s->current_frame >= s->nb_frames) {
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s->duration = pts;
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s->current_frame = 0;
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if (s->loop > 0)
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s->loop--;
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}
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return ret;
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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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AVFilterLink *outlink = ctx->outputs[0];
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LoopContext *s = ctx->priv;
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int ret = 0;
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if (inlink->frame_count_out >= s->start && s->size > 0 && s->loop != 0) {
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if (s->nb_frames < s->size) {
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if (!s->nb_frames)
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s->start_pts = frame->pts;
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s->frames[s->nb_frames] = av_frame_clone(frame);
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if (!s->frames[s->nb_frames]) {
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av_frame_free(&frame);
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return AVERROR(ENOMEM);
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}
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s->nb_frames++;
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s->duration = frame->pts + av_frame_get_pkt_duration(frame);
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ret = ff_filter_frame(outlink, frame);
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} else {
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av_frame_free(&frame);
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ret = push_frame(ctx);
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}
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} else {
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frame->pts += s->duration;
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ret = ff_filter_frame(outlink, frame);
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}
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return ret;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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LoopContext *s = ctx->priv;
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int ret = 0;
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if ((!s->size) ||
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(s->nb_frames < s->size) ||
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(s->nb_frames >= s->size && s->loop == 0)) {
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ret = ff_request_frame(ctx->inputs[0]);
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} else {
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ret = push_frame(ctx);
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}
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if (ret == AVERROR_EOF && s->nb_frames > 0 && s->loop != 0) {
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ret = push_frame(ctx);
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}
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return ret;
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}
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static const AVOption loop_options[] = {
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{ "loop", "number of loops", OFFSET(loop), AV_OPT_TYPE_INT, {.i64 = 0 }, -1, INT_MAX, VFLAGS },
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{ "size", "max number of frames to loop", OFFSET(size), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT16_MAX, VFLAGS },
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{ "start", "set the loop start frame", OFFSET(start), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, VFLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(loop);
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_VIDEO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_VIDEO,
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.request_frame = request_frame,
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},
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{ NULL }
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};
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AVFilter ff_vf_loop = {
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.name = "loop",
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.description = NULL_IF_CONFIG_SMALL("Loop video frames."),
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.priv_size = sizeof(LoopContext),
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.priv_class = &loop_class,
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.init = init,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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};
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#endif /* CONFIG_LOOP_FILTER */
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