lavfi: add ashowinfo filter
authorStefano Sabatini <stefano.sabatini-lala@poste.it>
Thu, 18 Aug 2011 16:19:55 +0000 (18:19 +0200)
committerStefano Sabatini <stefano.sabatini-lala@poste.it>
Sat, 20 Aug 2011 13:00:23 +0000 (15:00 +0200)
Useful for debugging.

Changelog
doc/filters.texi
libavfilter/Makefile
libavfilter/af_ashowinfo.c [new file with mode: 0644]
libavfilter/allfilters.c
libavfilter/avfilter.h

index c606eb2..61ea4e4 100644 (file)
--- 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:
index ce7d064..dd99c73 100644 (file)
@@ -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
index aa4654e..a3faf27 100644 (file)
@@ -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 (file)
index 0000000..25ec342
--- /dev/null
@@ -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}},
+};
index f623d00..5cf330c 100644 (file)
@@ -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);
 
index 4df18d2..af2d3cd 100644 (file)
@@ -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, \