lavfi: add volumedetect filter.

Changelog

@@ -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:

doc/filters.texi

@@ -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.

libavfilter/Makefile

@@ -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

libavfilter/af_volumedetect.c

+/*
+ * 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 }
+    },
+};

libavfilter/allfilters.c

@@ -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);