@@ -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

new file mode 100644

@@ -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}},

+};

@@ -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, \