[FFmpeg-cvslog] lavfi: remove astreamsync.
Nicolas George
git at videolan.org
Sat Nov 7 18:57:27 CET 2015
ffmpeg | branch: master | Nicolas George <george at nsup.org> | Sat Oct 24 16:13:32 2015 +0200| [d92e0848d9a5ad3c17253c13aaeada2817378609] | committer: Nicolas George
lavfi: remove astreamsync.
It was only useful for very specific testing purposes
and appears to be currently partially broken.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d92e0848d9a5ad3c17253c13aaeada2817378609
---
MAINTAINERS | 1 -
doc/filters.texi | 36 -------
libavfilter/Makefile | 1 -
libavfilter/af_astreamsync.c | 243 ------------------------------------------
libavfilter/allfilters.c | 1 -
5 files changed, 282 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index 96dab5e..3735742 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -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
diff --git a/doc/filters.texi b/doc/filters.texi
index a1147ff..f0b0ef3 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -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
- at section astreamsync
-
-Forward two audio streams and control the order the buffers are forwarded.
-
-The filter accepts the following options:
-
- at table @option
- at 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:
-
- at table @var
- at item b1 b2
-number of buffers forwarded so far on each stream
- at item s1 s2
-number of samples forwarded so far on each stream
- at item t1 t2
-current timestamp of each stream
- at end table
-
-The default value is @code{t1-t2}, which means to always forward the stream
-that has a smaller timestamp.
- at end table
-
- at subsection Examples
-
-Stress-test @code{amerge} by randomly sending buffers on the wrong
-input, while avoiding too much of a desynchronization:
- at 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
- at end example
-
@section asyncts
Synchronize audio data with timestamps by squeezing/stretching it and/or
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index db4f437..49b68db 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_astreamsync.c b/libavfilter/af_astreamsync.c
deleted file mode 100644
index d08da26..0000000
--- a/libavfilter/af_astreamsync.c
+++ /dev/null
@@ -1,243 +0,0 @@
-/*
- * Copyright (c) 2011 Nicolas George <nicolas.george at 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,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index e35c504..790587d 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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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