[FFmpeg-cvslog] lavfi: add ashowinfo filter
Stefano Sabatini
git at videolan.org
Sat Aug 20 15:10:42 CEST 2011
ffmpeg | branch: master | Stefano Sabatini <stefano.sabatini-lala at poste.it> | Thu Aug 18 18:19:55 2011 +0200| [e30a0b1b3b1f06bc34f43e8f77ddfd97503d8736] | committer: Stefano Sabatini
lavfi: add ashowinfo filter
Useful for debugging.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e30a0b1b3b1f06bc34f43e8f77ddfd97503d8736
---
Changelog | 1 +
doc/filters.texi | 50 ++++++++++++++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_ashowinfo.c | 99 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/avfilter.h | 2 +-
6 files changed, 153 insertions(+), 1 deletions(-)
diff --git a/Changelog b/Changelog
index c606eb2..61ea4e4 100644
--- a/Changelog
+++ b/Changelog
@@ -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:
diff --git a/doc/filters.texi b/doc/filters.texi
index ce7d064..dd99c73 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -137,6 +137,56 @@ For example, to resample the input audio to 44100Hz:
aresample=44100
@end example
+ at 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
+ at var{key}:@var{value}.
+
+A description of each shown parameter follows:
+
+ at table @option
+ at item n
+sequential number of the input frame, starting from 0
+
+ at 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}.
+
+ at item pts_time
+presentation TimeStamp of the input frame, expressed as a number of
+seconds
+
+ at 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)
+
+ at item fmt
+sample format name
+
+ at item chlayout
+channel layout description
+
+ at item nb_samples
+number of samples (per each channel) contained in the filtered frame
+
+ at item rate
+sample rate for the audio frame
+
+ at item planar
+if the packing format is planar, 0 if packed
+
+ at item checksum
+Adler-32 checksum of all the planes of the input frame
+
+ at 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}]"
+ at end table
+
@c man end AUDIO FILTERS
@chapter Audio Sources
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index aa4654e..a3faf27 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_ashowinfo.c b/libavfilter/af_ashowinfo.c
new file mode 100644
index 0000000..25ec342
--- /dev/null
+++ b/libavfilter/af_ashowinfo.c
@@ -0,0 +1,99 @@
+/*
+ * 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}},
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index f623d00..5cf330c 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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);
diff --git a/libavfilter/avfilter.h b/libavfilter/avfilter.h
index 4df18d2..af2d3cd 100644
--- a/libavfilter/avfilter.h
+++ b/libavfilter/avfilter.h
@@ -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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