@@ -50,6 +50,7 @@ version next:

 - edge detection filter

 - framestep filter

 - ffmpeg -shortest option is now per-output file

+- volume measurement filter

 version 0.11:

@@ -690,6 +690,46 @@ volume=-12dB

 @end example

 @end itemize

+@section volumedetect

+

+Detect the volume of the input video.

+

+The filter has no parameters. The input is not modified. Statistics about

+the volume will be printed in the log when the input stream end is reached.

+

+In particular it will show the mean volume (root mean square), maximum

+volume (on a per-sample basis), and the beginning of an histogram of the

+registered volume values (from the maximum value to a cumulated 1/1000 of

+the samples).

+

+All volumes are in decibels relative to the maximum PCM value.

+

+Here is an excerpt of the output:

+@example

+[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB

+[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB

+[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6

+[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62

+[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286

+[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042

+[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551

+[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609

+[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409

+@end example

+

+It means that:

+@itemize

+@item

+The mean square energy is approximately -27 dB, or 10^-2.7.

+@item

+The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.

+@item

+There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

+@end itemize

+

+In other words, raising the volume by +4 dB does not cause any clipping,

+raising it by +5 dB causes clipping for 6 samples, etc.

+

 @section asyncts

 Synchronize audio data with timestamps by squeezing/stretching it and/or

 dropping samples/adding silence when needed.

@@ -67,6 +67,7 @@ OBJS-$(CONFIG_PAN_FILTER)                    += af_pan.o

 OBJS-$(CONFIG_RESAMPLE_FILTER)               += af_resample.o

 OBJS-$(CONFIG_SILENCEDETECT_FILTER)          += af_silencedetect.o

 OBJS-$(CONFIG_VOLUME_FILTER)                 += af_volume.o

+OBJS-$(CONFIG_VOLUMEDETECT_FILTER)           += af_volumedetect.o

 OBJS-$(CONFIG_AEVALSRC_FILTER)               += asrc_aevalsrc.o

 OBJS-$(CONFIG_ANULLSRC_FILTER)               += asrc_anullsrc.o

new file mode 100644

@@ -0,0 +1,159 @@

+/*

+ * Copyright (c) 2012 Nicolas George

+ *

+ * 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

+ */

+

+#include "libavutil/audioconvert.h"

+#include "libavutil/avassert.h"

+#include "audio.h"

+#include "avfilter.h"

+#include "internal.h"

+

+typedef struct {

+    /**

+     * Number of samples at each PCM value.

+     * histogram[0x8000 + i] is the number of samples at value i.

+     * The extra element is there for symmetry.

+     */

+    uint64_t histogram[0x10001];

+} VolDetectContext;

+

+static int query_formats(AVFilterContext *ctx)

+{

+    enum AVSampleFormat sample_fmts[] = {

+        AV_SAMPLE_FMT_S16,

+        AV_SAMPLE_FMT_S16P,

+        AV_SAMPLE_FMT_NONE

+    };

+    AVFilterFormats *formats;

+

+    if (!(formats = ff_make_format_list(sample_fmts)))

+        return AVERROR(ENOMEM);

+    ff_set_common_formats(ctx, formats);

+

+    return 0;

+}

+

+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samples)

+{

+    AVFilterContext *ctx = inlink->dst;

+    VolDetectContext *vd = ctx->priv;

+    int64_t layout  = samples->audio->channel_layout;

+    int nb_samples  = samples->audio->nb_samples;

+    int nb_channels = av_get_channel_layout_nb_channels(layout);

+    int nb_planes   = nb_planes;

+    int plane, i;

+    int16_t *pcm;

+

+    if (!av_sample_fmt_is_planar(samples->format)) {

+        nb_samples *= nb_channels;

+        nb_planes = 1;

+    }

+    for (plane = 0; plane < nb_planes; plane++) {

+        pcm = (int16_t *)samples->extended_data[plane];

+        for (i = 0; i < nb_samples; i++)

+            vd->histogram[pcm[i] + 0x8000]++;

+    }

+

+    return ff_filter_samples(inlink->dst->outputs[0], samples);

+}

+

+#define MAX_DB 91

+

+static inline double logdb(uint64_t v)

+{

+    double d = v / (double)(0x8000 * 0x8000);

+    if (!v)

+        return MAX_DB;

+    return log(d) * -4.3429448190325182765112891891660508229; /* -10/log(10) */

+}

+

+static void print_stats(AVFilterContext *ctx)

+{

+    VolDetectContext *vd = ctx->priv;

+    int i, max_volume, shift;

+    uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;

+    uint64_t histdb[MAX_DB + 1] = { 0 };

+

+    for (i = 0; i < 0x10000; i++)

+        nb_samples += vd->histogram[i];

+    av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);

+    if (!nb_samples)

+        return;

+

+    /* If nb_samples > 1<<34, there is a risk of overflow in the

+       multiplication or the sum: shift all histogram values to avoid that.

+       The total number of samples must be recomputed to avoid rounding

+       errors. */

+    shift = av_log2(nb_samples >> 33);

+    for (i = 0; i < 0x10000; i++) {

+        nb_samples_shift += vd->histogram[i] >> shift;

+        power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);

+    }

+    if (!nb_samples_shift)

+        return;

+    power = (power + nb_samples_shift / 2) / nb_samples_shift;

+    av_assert0(power <= 0x8000 * 0x8000);

+    av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));

+

+    max_volume = 0x8000;

+    while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&

+                             !vd->histogram[0x8000 - max_volume])

+        max_volume--;

+    av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));

+

+    for (i = 0; i < 0x10000; i++)

+        histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];

+    for (i = 0; i <= MAX_DB && !histdb[i]; i++);

+    for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {

+        av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);

+        sum += histdb[i];

+    }

+}

+

+static int request_frame(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    int ret = ff_request_frame(ctx->inputs[0]);

+    if (ret == AVERROR_EOF)

+        print_stats(ctx);

+    return ret;

+}

+

+AVFilter avfilter_af_volumedetect = {

+    .name          = "volumedetect",

+    .description   = NULL_IF_CONFIG_SMALL("Detect audio volume."),

+

+    .priv_size     = sizeof(VolDetectContext),

+    .query_formats = query_formats,

+

+    .inputs    = (const AVFilterPad[]) {

+        { .name             = "default",

+          .type             = AVMEDIA_TYPE_AUDIO,

+          .get_audio_buffer = ff_null_get_audio_buffer,

+          .filter_samples   = filter_samples,

+          .min_perms        = AV_PERM_READ, },

+        { .name = NULL }

+    },

+    .outputs   = (const AVFilterPad[]) {

+        { .name = "default",

+          .type = AVMEDIA_TYPE_AUDIO,

+          .request_frame = request_frame, },

+        { .name = NULL }

+    },

+};

@@ -57,6 +57,7 @@ void avfilter_register_all(void)

     REGISTER_FILTER (PAN,         pan,         af);

     REGISTER_FILTER (SILENCEDETECT, silencedetect, af);

     REGISTER_FILTER (VOLUME,      volume,      af);

+    REGISTER_FILTER (VOLUMEDETECT,volumedetect,af);

     REGISTER_FILTER (RESAMPLE,    resample,    af);

     REGISTER_FILTER (AEVALSRC,    aevalsrc,    asrc);