[FFmpeg-cvslog] lavfi: add ashowinfo filter
Anton Khirnov
git at videolan.org
Tue Oct 30 15:16:39 CET 2012
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Tue Oct 23 21:37:26 2012 +0200| [20dd41af8513de427b00ee598339c9bc5778bdc5] | committer: Anton Khirnov
lavfi: add ashowinfo filter
It can be useful for debugging.
Based on a patch by Stefano Sabatini <stefano.sabatini-lala at poste.it>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=20dd41af8513de427b00ee598339c9bc5778bdc5
---
Changelog | 1 +
doc/filters.texi | 41 +++++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_ashowinfo.c | 136 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 181 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index c3d55c1..e2c4273 100644
--- a/Changelog
+++ b/Changelog
@@ -4,6 +4,7 @@ releases are sorted from youngest to oldest.
version <next>:
- metadata (INFO tag) support in WAV muxer
- support for building DLLs using MSVC
+- ashowinfo audio filter
version 9_beta1:
diff --git a/doc/filters.texi b/doc/filters.texi
index 8f90e84..85c8ae0 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -175,6 +175,47 @@ stream ends. The default value is 2 seconds.
Pass the audio source unchanged to the output.
+ 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, in time base units; the time base
+depends on the filter input pad, and is usually 1/@var{sample_rate}.
+
+ at item pts_time
+presentation timestamp of the input frame in seconds
+
+ at item fmt
+sample format
+
+ at item chlayout
+channel layout
+
+ at item rate
+sample rate for the audio frame
+
+ at item nb_samples
+number of samples (per channel) in the frame
+
+ at item checksum
+Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio
+the data is treated as if all the planes were concatenated.
+
+ at item plane_checksums
+A list of Adler-32 checksums for each data plane.
+ at end table
+
@section asplit
Split input audio into several identical outputs.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 530aa57..9770e1f 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -28,6 +28,7 @@ OBJS-$(CONFIG_AFIFO_FILTER) += fifo.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
+OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASYNCTS_FILTER) += af_asyncts.o
OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
diff --git a/libavfilter/af_ashowinfo.c b/libavfilter/af_ashowinfo.c
new file mode 100644
index 0000000..00e0322
--- /dev/null
+++ b/libavfilter/af_ashowinfo.c
@@ -0,0 +1,136 @@
+/*
+ * Copyright (c) 2011 Stefano Sabatini
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * filter for showing textual audio frame information
+ */
+
+#include <inttypes.h>
+#include <stddef.h>
+
+#include "libavutil/adler32.h"
+#include "libavutil/audioconvert.h"
+#include "libavutil/common.h"
+#include "libavutil/mem.h"
+#include "libavutil/samplefmt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+
+typedef struct AShowInfoContext {
+ /**
+ * Scratch space for individual plane checksums for planar audio
+ */
+ uint32_t *plane_checksums;
+
+ /**
+ * Frame counter
+ */
+ uint64_t frame;
+} AShowInfoContext;
+
+static int config_input(AVFilterLink *inlink)
+{
+ AShowInfoContext *s = inlink->dst->priv;
+ int channels = av_get_channel_layout_nb_channels(inlink->channel_layout);
+ s->plane_checksums = av_malloc(channels * sizeof(*s->plane_checksums));
+ if (!s->plane_checksums)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static void uninit(AVFilterContext *ctx)
+{
+ AShowInfoContext *s = ctx->priv;
+ av_freep(&s->plane_checksums);
+}
+
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AShowInfoContext *s = ctx->priv;
+ char chlayout_str[128];
+ uint32_t checksum = 0;
+ int channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
+ int planar = av_sample_fmt_is_planar(buf->format);
+ int block_align = av_get_bytes_per_sample(buf->format) * (planar ? 1 : channels);
+ int data_size = buf->audio->nb_samples * block_align;
+ int planes = planar ? channels : 1;
+ int i;
+
+ for (i = 0; i < planes; i++) {
+ uint8_t *data = buf->extended_data[i];
+
+ s->plane_checksums[i] = av_adler32_update(0, data, data_size);
+ checksum = i ? av_adler32_update(checksum, data, data_size) :
+ s->plane_checksums[0];
+ }
+
+ av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str), -1,
+ buf->audio->channel_layout);
+
+ av_log(ctx, AV_LOG_INFO,
+ "n:%"PRIu64" pts:%"PRId64" pts_time:%f "
+ "fmt:%s chlayout:%s rate:%d nb_samples:%d "
+ "checksum:%08X ",
+ s->frame, buf->pts, buf->pts * av_q2d(inlink->time_base),
+ av_get_sample_fmt_name(buf->format), chlayout_str,
+ buf->audio->sample_rate, buf->audio->nb_samples,
+ checksum);
+
+ av_log(ctx, AV_LOG_INFO, "plane_checksums: [ ");
+ for (i = 0; i < planes; i++)
+ av_log(ctx, AV_LOG_INFO, "%08X ", s->plane_checksums[i]);
+ av_log(ctx, AV_LOG_INFO, "]\n");
+
+ s->frame++;
+ return ff_filter_samples(inlink->dst->outputs[0], buf);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .get_audio_buffer = ff_null_get_audio_buffer,
+ .config_props = config_input,
+ .filter_samples = filter_samples,
+ .min_perms = AV_PERM_READ,
+ },
+ { NULL },
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL },
+};
+
+AVFilter avfilter_af_ashowinfo = {
+ .name = "ashowinfo",
+ .description = NULL_IF_CONFIG_SMALL("Show textual information for each audio frame."),
+ .priv_size = sizeof(AShowInfoContext),
+ .uninit = uninit,
+ .inputs = inputs,
+ .outputs = outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 94b3115..e759931 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -39,6 +39,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (AFORMAT, aformat, af);
REGISTER_FILTER (AMIX, amix, af);
REGISTER_FILTER (ANULL, anull, af);
+ REGISTER_FILTER (ASHOWINFO, ashowinfo, af);
REGISTER_FILTER (ASPLIT, asplit, af);
REGISTER_FILTER (ASYNCTS, asyncts, af);
REGISTER_FILTER (CHANNELMAP, channelmap, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 0e72a47..eb5326b 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
-#define LIBAVFILTER_VERSION_MINOR 1
+#define LIBAVFILTER_VERSION_MINOR 2
#define LIBAVFILTER_VERSION_MICRO 0
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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