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FFmpeg/libavfilter/af_astreamsync.c

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/*
* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
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*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Stream (de)synchronization filter
*/
#include "libavutil/eval.h"
#include "avfilter.h"
#include "audio.h"
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#include "internal.h"
#define QUEUE_SIZE 16
static const char * const var_names[] = {
"b1", "b2",
"s1", "s2",
"t1", "t2",
NULL
};
enum var_name {
VAR_B1, VAR_B2,
VAR_S1, VAR_S2,
VAR_T1, VAR_T2,
VAR_NB
};
typedef struct {
AVExpr *expr;
double var_values[VAR_NB];
struct buf_queue {
AVFilterBufferRef *buf[QUEUE_SIZE];
unsigned tail, nb;
/* buf[tail] is the oldest,
buf[(tail + nb) % QUEUE_SIZE] is where the next is added */
} queue[2];
int req[2];
int next_out;
int eof; /* bitmask, one bit for each stream */
} AStreamSyncContext;
static const char *default_expr = "t1-t2";
static av_cold int init(AVFilterContext *ctx, const char *args0)
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{
AStreamSyncContext *as = ctx->priv;
const char *expr = args0 ? args0 : default_expr;
int r, i;
r = av_expr_parse(&as->expr, expr, var_names,
NULL, NULL, NULL, NULL, 0, ctx);
if (r < 0) {
av_log(ctx, AV_LOG_ERROR, "Error in expression \"%s\"\n", expr);
return r;
}
for (i = 0; i < 42; i++)
av_expr_eval(as->expr, as->var_values, NULL); /* exercize prng */
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
int i;
AVFilterFormats *formats, *rates;
Merge remote-tracking branch 'qatar/master' * qatar/master: (26 commits) fate: use diff -b in oneline comparison Add missing version bumps and APIchanges/Changelog entries. lavfi: move buffer management function to a separate file. lavfi: move formats-related functions from default.c to formats.c lavfi: move video-related functions to a separate file. fate: make smjpeg a demux test fate: separate sierra-vmd audio and video tests fate: separate smacker audio and video tests libmp3lame: set supported channel layouts. avconv: automatically insert asyncts when -async is used. avconv: add support for audio filters. lavfi: add asyncts filter. lavfi: add aformat filter lavfi: add an audio buffer sink. lavfi: add an audio buffer source. buffersrc: add av_buffersrc_write_frame(). buffersrc: fix invalid read in uninit if the fifo hasn't been allocated lavfi: rename vsrc_buffer.c to buffersrc.c avfiltergraph: reindent lavfi: add channel layout/sample rate negotiation. ... Conflicts: Changelog doc/APIchanges doc/filters.texi ffmpeg.c ffprobe.c libavcodec/libmp3lame.c libavfilter/Makefile libavfilter/af_aformat.c libavfilter/allfilters.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/avfiltergraph.c libavfilter/buffersrc.c libavfilter/defaults.c libavfilter/formats.c libavfilter/src_buffer.c libavfilter/version.h libavfilter/vf_yadif.c libavfilter/vsrc_buffer.c libavfilter/vsrc_buffer.h libavutil/avutil.h tests/fate/audio.mak tests/fate/demux.mak tests/fate/video.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-16 03:27:31 +03:00
AVFilterChannelLayouts *layouts;
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for (i = 0; i < 2; i++) {
formats = ctx->inputs[i]->in_formats;
ff_formats_ref(formats, &ctx->inputs[i]->out_formats);
ff_formats_ref(formats, &ctx->outputs[i]->in_formats);
rates = ff_all_samplerates();
ff_formats_ref(rates, &ctx->inputs[i]->out_samplerates);
ff_formats_ref(rates, &ctx->outputs[i]->in_samplerates);
Merge remote-tracking branch 'qatar/master' * qatar/master: (26 commits) fate: use diff -b in oneline comparison Add missing version bumps and APIchanges/Changelog entries. lavfi: move buffer management function to a separate file. lavfi: move formats-related functions from default.c to formats.c lavfi: move video-related functions to a separate file. fate: make smjpeg a demux test fate: separate sierra-vmd audio and video tests fate: separate smacker audio and video tests libmp3lame: set supported channel layouts. avconv: automatically insert asyncts when -async is used. avconv: add support for audio filters. lavfi: add asyncts filter. lavfi: add aformat filter lavfi: add an audio buffer sink. lavfi: add an audio buffer source. buffersrc: add av_buffersrc_write_frame(). buffersrc: fix invalid read in uninit if the fifo hasn't been allocated lavfi: rename vsrc_buffer.c to buffersrc.c avfiltergraph: reindent lavfi: add channel layout/sample rate negotiation. ... Conflicts: Changelog doc/APIchanges doc/filters.texi ffmpeg.c ffprobe.c libavcodec/libmp3lame.c libavfilter/Makefile libavfilter/af_aformat.c libavfilter/allfilters.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/avfiltergraph.c libavfilter/buffersrc.c libavfilter/defaults.c libavfilter/formats.c libavfilter/src_buffer.c libavfilter/version.h libavfilter/vf_yadif.c libavfilter/vsrc_buffer.c libavfilter/vsrc_buffer.h libavutil/avutil.h tests/fate/audio.mak tests/fate/demux.mak tests/fate/video.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
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layouts = ctx->inputs[i]->in_channel_layouts;
ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts);
ff_channel_layouts_ref(layouts, &ctx->outputs[i]->in_channel_layouts);
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}
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
int id = outlink == ctx->outputs[1];
outlink->sample_rate = ctx->inputs[id]->sample_rate;
outlink->time_base = ctx->inputs[id]->time_base;
return 0;
}
static int send_out(AVFilterContext *ctx, int out_id)
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{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFilterBufferRef *buf = queue->buf[queue->tail];
int ret;
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queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
as->var_values[VAR_S1 + out_id] += buf->audio->nb_samples;
if (buf->pts != AV_NOPTS_VALUE)
as->var_values[VAR_T1 + out_id] =
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
ret = ff_filter_samples(ctx->outputs[out_id], buf);
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queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
return ret;
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}
static void send_next(AVFilterContext *ctx)
{
AStreamSyncContext *as = ctx->priv;
int i;
while (1) {
if (!as->queue[as->next_out].nb)
break;
send_out(ctx, as->next_out);
if (!as->eof)
as->next_out = av_expr_eval(as->expr, as->var_values, NULL) >= 0;
}
for (i = 0; i < 2; i++)
if (as->queue[i].nb == QUEUE_SIZE)
send_out(ctx, i);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AStreamSyncContext *as = ctx->priv;
int id = outlink == ctx->outputs[1];
as->req[id]++;
while (as->req[id] && !(as->eof & (1 << id))) {
if (as->queue[as->next_out].nb) {
send_next(ctx);
} else {
as->eof |= 1 << as->next_out;
ff_request_frame(ctx->inputs[as->next_out]);
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if (as->eof & (1 << as->next_out))
as->next_out = !as->next_out;
}
}
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
AVFilterContext *ctx = inlink->dst;
AStreamSyncContext *as = ctx->priv;
int id = inlink == ctx->inputs[1];
as->queue[id].buf[(as->queue[id].tail + as->queue[id].nb++) % QUEUE_SIZE] =
insamples;
as->eof &= ~(1 << id);
send_next(ctx);
return 0;
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}
AVFilter avfilter_af_astreamsync = {
.name = "astreamsync",
.description = NULL_IF_CONFIG_SMALL("Copy two streams of audio data "
"in a configurable order."),
.priv_size = sizeof(AStreamSyncContext),
.init = init,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {
{ .name = "in1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = "in2",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{ .name = "out1",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame, },
{ .name = "out2",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame, },
{ .name = NULL }
},
};