lavfi: add astreamsync audio filter.
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@ -12,6 +12,7 @@ version next:
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- XML output in ffprobe
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- asplit audio filter
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- tinterlace video filter
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- astreamsync audio filter
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version 0.9:
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@ -237,6 +237,35 @@ For example:
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will create two separate outputs from the same input, one cropped and
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one padded.
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@section astreamsync
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Forward two audio streams and control the order the buffers are forwarded.
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The argument to the filter is an expression deciding which stream should be
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forwarded next: if the result is negative, the first stream is forwarded; if
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the result is positive or zero, the second stream is forwarded. It can use
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the following variables:
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@table @var
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@item b1 b2
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number of buffers forwarded so far on each stream
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@item s1 s2
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number of samples forwarded so far on each stream
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@item t1 t2
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current timestamp of each stream
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@end table
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The default value is @code{t1-t2}, which means to always forward the stream
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that has a smaller timestamp.
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Example: stress-test @code{amerge} by randomly sending buffers on the wrong
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input, while avoiding too much of a desynchronization:
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@example
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amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
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[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
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[a2] [b2] amerge
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@end example
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@section earwax
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Make audio easier to listen to on headphones.
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@ -30,6 +30,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
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OBJS-$(CONFIG_ASPLIT_FILTER) += af_asplit.o
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OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
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OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
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OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
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OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
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209
libavfilter/af_astreamsync.c
Normal file
209
libavfilter/af_astreamsync.c
Normal file
@ -0,0 +1,209 @@
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/*
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* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
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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
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* GNU 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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* Stream (de)synchronization filter
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*/
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#include "libavutil/eval.h"
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#include "avfilter.h"
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#include "internal.h"
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#define QUEUE_SIZE 16
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static const char * const var_names[] = {
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"b1", "b2",
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"s1", "s2",
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"t1", "t2",
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NULL
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};
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enum var_name {
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VAR_B1, VAR_B2,
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VAR_S1, VAR_S2,
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VAR_T1, VAR_T2,
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VAR_NB
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};
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typedef struct {
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AVExpr *expr;
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double var_values[VAR_NB];
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struct buf_queue {
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AVFilterBufferRef *buf[QUEUE_SIZE];
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unsigned tail, nb;
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/* buf[tail] is the oldest,
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buf[(tail + nb) % QUEUE_SIZE] is where the next is added */
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} queue[2];
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int req[2];
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int next_out;
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int eof; /* bitmask, one bit for each stream */
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} AStreamSyncContext;
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static const char *default_expr = "t1-t2";
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static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
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{
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AStreamSyncContext *as = ctx->priv;
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const char *expr = args0 ? args0 : default_expr;
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int r, i;
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r = av_expr_parse(&as->expr, expr, var_names,
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NULL, NULL, NULL, NULL, 0, ctx);
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if (r < 0) {
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av_log(ctx, AV_LOG_ERROR, "Error in expression \"%s\"\n", expr);
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return r;
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}
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for (i = 0; i < 42; i++)
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av_expr_eval(as->expr, as->var_values, NULL); /* exercize prng */
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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int i;
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AVFilterFormats *formats;
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for (i = 0; i < 2; i++) {
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formats = ctx->inputs[i]->in_formats;
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avfilter_formats_ref(formats, &ctx->inputs[i]->out_formats);
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avfilter_formats_ref(formats, &ctx->outputs[i]->in_formats);
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formats = ctx->inputs[i]->in_packing;
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avfilter_formats_ref(formats, &ctx->inputs[i]->out_packing);
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avfilter_formats_ref(formats, &ctx->outputs[i]->in_packing);
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formats = ctx->inputs[i]->in_chlayouts;
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avfilter_formats_ref(formats, &ctx->inputs[i]->out_chlayouts);
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avfilter_formats_ref(formats, &ctx->outputs[i]->in_chlayouts);
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}
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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int id = outlink == ctx->outputs[1];
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int i;
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outlink->sample_rate = ctx->inputs[id]->sample_rate;
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outlink->time_base = ctx->inputs[id]->time_base;
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return 0;
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}
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static void send_out(AVFilterContext *ctx, int out_id)
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{
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AStreamSyncContext *as = ctx->priv;
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struct buf_queue *queue = &as->queue[out_id];
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AVFilterBufferRef *buf = queue->buf[queue->tail];
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queue->buf[queue->tail] = NULL;
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as->var_values[VAR_B1 + out_id]++;
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as->var_values[VAR_S1 + out_id] += buf->audio->nb_samples;
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if (buf->pts != AV_NOPTS_VALUE)
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as->var_values[VAR_T1 + out_id] =
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av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
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as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
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(double)ctx->inputs[out_id]->sample_rate;
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avfilter_filter_samples(ctx->outputs[out_id], buf);
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queue->nb--;
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queue->tail = (queue->tail + 1) % QUEUE_SIZE;
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if (as->req[out_id])
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as->req[out_id]--;
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}
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static void send_next(AVFilterContext *ctx)
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{
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AStreamSyncContext *as = ctx->priv;
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int i;
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while (1) {
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if (!as->queue[as->next_out].nb)
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break;
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send_out(ctx, as->next_out);
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if (!as->eof)
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as->next_out = av_expr_eval(as->expr, as->var_values, NULL) >= 0;
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}
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for (i = 0; i < 2; i++)
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if (as->queue[i].nb == QUEUE_SIZE)
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send_out(ctx, i);
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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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AStreamSyncContext *as = ctx->priv;
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int id = outlink == ctx->outputs[1];
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as->req[id]++;
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while (as->req[id] && !(as->eof & (1 << id))) {
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if (as->queue[as->next_out].nb) {
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send_next(ctx);
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} else {
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as->eof |= 1 << as->next_out;
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avfilter_request_frame(ctx->inputs[as->next_out]);
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if (as->eof & (1 << as->next_out))
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as->next_out = !as->next_out;
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}
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}
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return 0;
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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AStreamSyncContext *as = ctx->priv;
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int id = inlink == ctx->inputs[1];
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as->queue[id].buf[(as->queue[id].tail + as->queue[id].nb++) % QUEUE_SIZE] =
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insamples;
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as->eof &= ~(1 << id);
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send_next(ctx);
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}
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AVFilter avfilter_af_astreamsync = {
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.name = "astreamsync",
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.description = NULL_IF_CONFIG_SMALL("Copy two streams of audio data "
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"in a configurable order."),
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.priv_size = sizeof(AStreamSyncContext),
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.init = init,
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.query_formats = query_formats,
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.inputs = (const AVFilterPad[]) {
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{ .name = "in1",
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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 = "in2",
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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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},
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.outputs = (const AVFilterPad[]) {
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{ .name = "out1",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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.request_frame = request_frame, },
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{ .name = "out2",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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.request_frame = request_frame, },
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{ .name = NULL }
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},
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};
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@ -40,6 +40,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER (ARESAMPLE, aresample, af);
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REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
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REGISTER_FILTER (ASPLIT, asplit, af);
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REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);
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REGISTER_FILTER (EARWAX, earwax, af);
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REGISTER_FILTER (PAN, pan, af);
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REGISTER_FILTER (VOLUME, volume, af);
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