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lavfi: remove astreamsync. 

It was only useful for very specific testing purposes
and appears to be currently partially broken.
tags/n3.0
Nicolas George 10 years ago
parent
47ea04ff10
commit
d92e0848d9
5 changed files with 0 additions and 282 deletions
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  1. +0
    -1
      MAINTAINERS
  2. +0
    -36
      doc/filters.texi
  3. +0
    -1
      libavfilter/Makefile
  4. +0
    -243
      libavfilter/af_astreamsync.c
  5. +0
    -1
      libavfilter/allfilters.c

+ 0
- 1
MAINTAINERS View File

@@ -347,7 +347,6 @@ Filters:
af_aphaser.c Paul B Mahol
af_aresample.c Michael Niedermayer
af_astats.c Paul B Mahol
af_astreamsync.c Nicolas George
af_atempo.c Pavel Koshevoy
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol


+ 0
- 36
doc/filters.texi View File

@@ -1170,42 +1170,6 @@ Number of occasions (not the number of samples) that the signal attained either
Overall bit depth of audio. Number of bits used for each sample.
@end table

@section astreamsync

Forward two audio streams and control the order the buffers are forwarded.

The filter accepts the following options:

@table @option
@item expr, e
Set the expression deciding which stream should be
forwarded next: if the result is negative, the first stream is forwarded; if
the result is positive or zero, the second stream is forwarded. It can use
the following variables:

@table @var
@item b1 b2
number of buffers forwarded so far on each stream
@item s1 s2
number of samples forwarded so far on each stream
@item t1 t2
current timestamp of each stream
@end table

The default value is @code{t1-t2}, which means to always forward the stream
that has a smaller timestamp.
@end table

@subsection Examples

Stress-test @code{amerge} by randomly sending buffers on the wrong
input, while avoiding too much of a desynchronization:
@example
amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
[a2] [b2] amerge
@end example

@section asyncts

Synchronize audio data with timestamps by squeezing/stretching it and/or


+ 0
- 1
libavfilter/Makefile View File

@@ -51,7 +51,6 @@ OBJS-$(CONFIG_ASETTB_FILTER) += settb.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
OBJS-$(CONFIG_ATRIM_FILTER) += trim.o


+ 0
- 243
libavfilter/af_astreamsync.c View File

@@ -1,243 +0,0 @@
/*
* 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.
*
* 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 "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#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 {
const AVClass *class;
AVExpr *expr;
char *expr_str;
double var_values[VAR_NB];
struct buf_queue {
AVFrame *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;

#define OFFSET(x) offsetof(AStreamSyncContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption astreamsync_options[] = {
{ "expr", "set stream selection expression", OFFSET(expr_str), AV_OPT_TYPE_STRING, { .str = "t1-t2" }, .flags = FLAGS },
{ "e", "set stream selection expression", OFFSET(expr_str), AV_OPT_TYPE_STRING, { .str = "t1-t2" }, .flags = FLAGS },
{ NULL }
};

AVFILTER_DEFINE_CLASS(astreamsync);

static av_cold int init(AVFilterContext *ctx)
{
AStreamSyncContext *as = ctx->priv;
int r, i;

r = av_expr_parse(&as->expr, as->expr_str, var_names,
NULL, NULL, NULL, NULL, 0, ctx);
if (r < 0) {
av_log(ctx, AV_LOG_ERROR, "Error in expression \"%s\"\n", as->expr_str);
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, ret;
AVFilterFormats *formats, *rates;
AVFilterChannelLayouts *layouts;

for (i = 0; i < 2; i++) {
formats = ctx->inputs[i]->in_formats;
if ((ret = ff_formats_ref(formats, &ctx->inputs[i]->out_formats)) < 0 ||
(ret = ff_formats_ref(formats, &ctx->outputs[i]->in_formats)) < 0)
return ret;
rates = ff_all_samplerates();
if ((ret = ff_formats_ref(rates, &ctx->inputs[i]->out_samplerates)) < 0 ||
(ret = ff_formats_ref(rates, &ctx->outputs[i]->in_samplerates)) < 0)
return ret;
layouts = ctx->inputs[i]->in_channel_layouts;
if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0 ||
(ret = ff_channel_layouts_ref(layouts, &ctx->outputs[i]->in_channel_layouts)) < 0)
return ret;
}
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)
{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFrame *buf = queue->buf[queue->tail];
int ret;

queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
as->var_values[VAR_S1 + out_id] += buf->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->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
ret = ff_filter_frame(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
return ret;
}

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]);
if (as->eof & (1 << as->next_out))
as->next_out = !as->next_out;
}
}
return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
{
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;
}

static av_cold void uninit(AVFilterContext *ctx)
{
AStreamSyncContext *as = ctx->priv;

av_expr_free(as->expr);
as->expr = NULL;
}

static const AVFilterPad astreamsync_inputs[] = {
{
.name = "in1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},{
.name = "in2",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};

static const AVFilterPad astreamsync_outputs[] = {
{
.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,
},
{ NULL }
};

AVFilter ff_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,
.uninit = uninit,
.query_formats = query_formats,
.inputs = astreamsync_inputs,
.outputs = astreamsync_outputs,
.priv_class = &astreamsync_class,
};

+ 0
- 1
libavfilter/allfilters.c View File

@@ -73,7 +73,6 @@ void avfilter_register_all(void)
REGISTER_FILTER(ASHOWINFO, ashowinfo, af);
REGISTER_FILTER(ASPLIT, asplit, af);
REGISTER_FILTER(ASTATS, astats, af);
REGISTER_FILTER(ASTREAMSYNC, astreamsync, af);
REGISTER_FILTER(ASYNCTS, asyncts, af);
REGISTER_FILTER(ATEMPO, atempo, af);
REGISTER_FILTER(ATRIM, atrim, af);


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