Commit e30a0b1b authored by Stefano Sabatini's avatar Stefano Sabatini

lavfi: add ashowinfo filter

Useful for debugging.
parent 9d84a17b
......@@ -44,6 +44,7 @@ easier to use. The changes are:
* -intra option was removed, it's equivalent to -g 0.
- XMV demuxer
- LOAS demuxer
- ashowinfo filter added
version 0.8:
......
......@@ -137,6 +137,56 @@ For example, to resample the input audio to 44100Hz:
aresample=44100
@end example
@section ashowinfo
Show a line containing various information for each input audio frame.
The input audio is not modified.
The shown line contains a sequence of key/value pairs of the form
@var{key}:@var{value}.
A description of each shown parameter follows:
@table @option
@item n
sequential number of the input frame, starting from 0
@item pts
presentation TimeStamp of the input frame, expressed as a number of
time base units. The time base unit depends on the filter input pad, and
is usually 1/@var{sample_rate}.
@item pts_time
presentation TimeStamp of the input frame, expressed as a number of
seconds
@item pos
position of the frame in the input stream, -1 if this information in
unavailable and/or meanigless (for example in case of synthetic audio)
@item fmt
sample format name
@item chlayout
channel layout description
@item nb_samples
number of samples (per each channel) contained in the filtered frame
@item rate
sample rate for the audio frame
@item planar
if the packing format is planar, 0 if packed
@item checksum
Adler-32 checksum of all the planes of the input frame
@item plane_checksum
Adler-32 checksum for each input frame plane, expressed in the form
"[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5} @var{c6} @var{c7}]"
@end table
@c man end AUDIO FILTERS
@chapter Audio Sources
......
......@@ -22,6 +22,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
......
/*
* Copyright (c) 2011 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
* filter fow showing textual audio frame information
*/
#include "libavutil/adler32.h"
#include "libavutil/audioconvert.h"
#include "avfilter.h"
typedef struct {
unsigned int frame;
} ShowInfoContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
ShowInfoContext *showinfo = ctx->priv;
showinfo->frame = 0;
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
ShowInfoContext *showinfo = ctx->priv;
uint32_t plane_checksum[8] = {0}, checksum = 0;
char chlayout_str[128];
int plane;
for (plane = 0; samplesref->data[plane] && plane < 8; plane++) {
uint8_t *data = samplesref->data[plane];
int linesize = samplesref->linesize[plane];
plane_checksum[plane] = av_adler32_update(plane_checksum[plane],
data, linesize);
checksum = av_adler32_update(checksum, data, linesize);
}
av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
samplesref->audio->channel_layout);
av_log(ctx, AV_LOG_INFO,
"n:%d pts:%"PRId64" pts_time:%f pos:%"PRId64" "
"fmt:%s chlayout:%s nb_samples:%d rate:%d planar:%d "
"checksum:%u plane_checksum[%u %u %u %u %u %u %u %u]\n",
showinfo->frame,
samplesref->pts, samplesref->pts * av_q2d(inlink->time_base),
samplesref->pos,
av_get_sample_fmt_name(samplesref->format),
chlayout_str,
samplesref->audio->nb_samples,
samplesref->audio->sample_rate,
samplesref->audio->planar,
checksum,
plane_checksum[0], plane_checksum[1], plane_checksum[2], plane_checksum[3],
plane_checksum[4], plane_checksum[5], plane_checksum[6], plane_checksum[7]);
showinfo->frame++;
avfilter_filter_samples(inlink->dst->outputs[0], samplesref);
}
AVFilter avfilter_af_ashowinfo = {
.name = "ashowinfo",
.description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
.priv_size = sizeof(ShowInfoContext),
.init = init,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = avfilter_null_get_audio_buffer,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO },
{ .name = NULL}},
};
......@@ -37,6 +37,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (ANULL, anull, af);
REGISTER_FILTER (ARESAMPLE, aresample, af);
REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
......
......@@ -29,7 +29,7 @@
#include "libavutil/rational.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 32
#define LIBAVFILTER_VERSION_MINOR 33
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
......
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