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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavfilter/f_settb.c
Michael Niedermayer f8911b987d Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-09 22:40:12 +02:00

186 lines
5.8 KiB
C

/*
* Copyright (c) 2010 Stefano Sabatini
*
* 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
* Set timebase for the output link.
*/
#include "libavutil/avstring.h"
#include "libavutil/eval.h"
#include "libavutil/mathematics.h"
#include "libavutil/rational.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#include "video.h"
static const char *const var_names[] = {
"AVTB", /* default timebase 1/AV_TIME_BASE */
"intb", /* input timebase */
"sr", /* sample rate */
NULL
};
enum var_name {
VAR_AVTB,
VAR_INTB,
VAR_SR,
VAR_VARS_NB
};
typedef struct {
char tb_expr[256];
double var_values[VAR_VARS_NB];
} SetTBContext;
static av_cold int init(AVFilterContext *ctx, const char *args)
{
SetTBContext *settb = ctx->priv;
av_strlcpy(settb->tb_expr, "intb", sizeof(settb->tb_expr));
if (args)
sscanf(args, "%255[^:]", settb->tb_expr);
return 0;
}
static int config_output_props(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SetTBContext *settb = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVRational time_base;
int ret;
double res;
settb->var_values[VAR_AVTB] = av_q2d(AV_TIME_BASE_Q);
settb->var_values[VAR_INTB] = av_q2d(inlink->time_base);
settb->var_values[VAR_SR] = inlink->sample_rate;
outlink->w = inlink->w;
outlink->h = inlink->h;
if ((ret = av_expr_parse_and_eval(&res, settb->tb_expr, var_names, settb->var_values,
NULL, NULL, NULL, NULL, NULL, 0, NULL)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid expression '%s' for timebase.\n", settb->tb_expr);
return ret;
}
time_base = av_d2q(res, INT_MAX);
if (time_base.num <= 0 || time_base.den <= 0) {
av_log(ctx, AV_LOG_ERROR,
"Invalid non-positive values for the timebase num:%d or den:%d.\n",
time_base.num, time_base.den);
return AVERROR(EINVAL);
}
outlink->time_base = time_base;
av_log(outlink->src, AV_LOG_VERBOSE, "tb:%d/%d -> tb:%d/%d\n",
inlink ->time_base.num, inlink ->time_base.den,
outlink->time_base.num, outlink->time_base.den);
return 0;
}
static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterBufferRef *picref2 = picref;
if (av_cmp_q(inlink->time_base, outlink->time_base)) {
picref2 = avfilter_ref_buffer(picref, ~0);
picref2->pts = av_rescale_q(picref->pts, inlink->time_base, outlink->time_base);
av_log(ctx, AV_LOG_DEBUG, "tb:%d/%d pts:%"PRId64" -> tb:%d/%d pts:%"PRId64"\n",
inlink ->time_base.num, inlink ->time_base.den, picref ->pts,
outlink->time_base.num, outlink->time_base.den, picref2->pts);
avfilter_unref_buffer(picref);
}
ff_start_frame(outlink, picref2);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterBufferRef *outsamples = insamples;
if (av_cmp_q(inlink->time_base, outlink->time_base)) {
outsamples = avfilter_ref_buffer(insamples, ~0);
outsamples->pts = av_rescale_q(insamples->pts, inlink->time_base, outlink->time_base);
av_log(ctx, AV_LOG_DEBUG, "tb:%d/%d pts:%"PRId64" -> tb:%d/%d pts:%"PRId64"\n",
inlink ->time_base.num, inlink ->time_base.den, insamples ->pts,
outlink->time_base.num, outlink->time_base.den, outsamples->pts);
avfilter_unref_buffer(insamples);
}
return ff_filter_samples(outlink, outsamples);
}
#if CONFIG_SETTB_FILTER
AVFilter avfilter_vf_settb = {
.name = "settb",
.description = NULL_IF_CONFIG_SMALL("Set timebase for the video output link."),
.init = init,
.priv_size = sizeof(SetTBContext),
.inputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.get_video_buffer = ff_null_get_video_buffer,
.start_frame = start_frame,
.end_frame = ff_null_end_frame },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_output_props, },
{ .name = NULL}
},
};
#endif
#if CONFIG_ASETTB_FILTER
AVFilter avfilter_af_asettb = {
.name = "asettb",
.description = NULL_IF_CONFIG_SMALL("Set timebase for the audio output link."),
.init = init,
.priv_size = sizeof(SetTBContext),
.inputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output_props, },
{ .name = NULL}
},
};
#endif