lavfi: add ashowinfo filter
Useful for debugging.
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@ -44,6 +44,7 @@ easier to use. The changes are:
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* -intra option was removed, it's equivalent to -g 0.
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- XMV demuxer
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- LOAS demuxer
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- ashowinfo filter added
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version 0.8:
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@ -137,6 +137,56 @@ For example, to resample the input audio to 44100Hz:
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aresample=44100
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@end example
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@section ashowinfo
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Show a line containing various information for each input audio frame.
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The input audio is not modified.
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The shown line contains a sequence of key/value pairs of the form
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@var{key}:@var{value}.
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A description of each shown parameter follows:
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@table @option
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@item n
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sequential number of the input frame, starting from 0
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@item pts
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presentation TimeStamp of the input frame, expressed as a number of
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time base units. The time base unit depends on the filter input pad, and
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is usually 1/@var{sample_rate}.
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@item pts_time
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presentation TimeStamp of the input frame, expressed as a number of
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seconds
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@item pos
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position of the frame in the input stream, -1 if this information in
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unavailable and/or meanigless (for example in case of synthetic audio)
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@item fmt
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sample format name
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@item chlayout
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channel layout description
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@item nb_samples
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number of samples (per each channel) contained in the filtered frame
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@item rate
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sample rate for the audio frame
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@item planar
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if the packing format is planar, 0 if packed
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@item checksum
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Adler-32 checksum of all the planes of the input frame
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@item plane_checksum
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Adler-32 checksum for each input frame plane, expressed in the form
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"[@var{c0} @var{c1} @var{c2} @var{c3} @var{c4} @var{c5} @var{c6} @var{c7}]"
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@end table
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@c man end AUDIO FILTERS
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@chapter Audio Sources
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@ -22,6 +22,7 @@ OBJS-$(CONFIG_AVCODEC) += avcodec.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
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OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
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OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
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99
libavfilter/af_ashowinfo.c
Normal file
99
libavfilter/af_ashowinfo.c
Normal file
@ -0,0 +1,99 @@
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/*
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* Copyright (c) 2011 Stefano Sabatini
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* filter fow showing textual audio frame information
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*/
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#include "libavutil/adler32.h"
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#include "libavutil/audioconvert.h"
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#include "avfilter.h"
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typedef struct {
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unsigned int frame;
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} ShowInfoContext;
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static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
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{
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ShowInfoContext *showinfo = ctx->priv;
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showinfo->frame = 0;
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return 0;
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
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{
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AVFilterContext *ctx = inlink->dst;
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ShowInfoContext *showinfo = ctx->priv;
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uint32_t plane_checksum[8] = {0}, checksum = 0;
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char chlayout_str[128];
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int plane;
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for (plane = 0; samplesref->data[plane] && plane < 8; plane++) {
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uint8_t *data = samplesref->data[plane];
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int linesize = samplesref->linesize[plane];
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plane_checksum[plane] = av_adler32_update(plane_checksum[plane],
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data, linesize);
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checksum = av_adler32_update(checksum, data, linesize);
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}
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av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
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samplesref->audio->channel_layout);
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av_log(ctx, AV_LOG_INFO,
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"n:%d pts:%"PRId64" pts_time:%f pos:%"PRId64" "
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"fmt:%s chlayout:%s nb_samples:%d rate:%d planar:%d "
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"checksum:%u plane_checksum[%u %u %u %u %u %u %u %u]\n",
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showinfo->frame,
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samplesref->pts, samplesref->pts * av_q2d(inlink->time_base),
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samplesref->pos,
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av_get_sample_fmt_name(samplesref->format),
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chlayout_str,
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samplesref->audio->nb_samples,
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samplesref->audio->sample_rate,
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samplesref->audio->planar,
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checksum,
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plane_checksum[0], plane_checksum[1], plane_checksum[2], plane_checksum[3],
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plane_checksum[4], plane_checksum[5], plane_checksum[6], plane_checksum[7]);
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showinfo->frame++;
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avfilter_filter_samples(inlink->dst->outputs[0], samplesref);
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}
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AVFilter avfilter_af_ashowinfo = {
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.name = "ashowinfo",
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.description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
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.priv_size = sizeof(ShowInfoContext),
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.init = init,
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.get_audio_buffer = avfilter_null_get_audio_buffer,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO },
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{ .name = NULL}},
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};
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@ -37,6 +37,7 @@ void avfilter_register_all(void)
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REGISTER_FILTER (AFORMAT, aformat, af);
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REGISTER_FILTER (ANULL, anull, af);
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REGISTER_FILTER (ARESAMPLE, aresample, af);
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REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
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REGISTER_FILTER (ANULLSRC, anullsrc, asrc);
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@ -29,7 +29,7 @@
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#include "libavutil/rational.h"
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#define LIBAVFILTER_VERSION_MAJOR 2
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#define LIBAVFILTER_VERSION_MINOR 32
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#define LIBAVFILTER_VERSION_MINOR 33
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#define LIBAVFILTER_VERSION_MICRO 0
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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