Add audiolevel filter.

src/modules/core/factory.c

@@ -75,8 +75,8 @@ MLT_REPOSITORY
 	MLT_REGISTER( filter_type, "audiowave", filter_audiowave_init );
 	MLT_REGISTER( filter_type, "brightness", filter_brightness_init );
 	MLT_REGISTER( filter_type, "channelcopy", filter_channelcopy_init );
-    MLT_REGISTER( filter_type, "channelswap", filter_channelcopy_init );
-    MLT_REGISTER( filter_type, "crop", filter_crop_init );
+	MLT_REGISTER( filter_type, "channelswap", filter_channelcopy_init );
+	MLT_REGISTER( filter_type, "crop", filter_crop_init );
 	MLT_REGISTER( filter_type, "data_feed", filter_data_feed_init );
 	MLT_REGISTER( filter_type, "data_show", filter_data_show_init );
 	MLT_REGISTER( filter_type, "fieldorder", filter_fieldorder_init );

src/modules/normalize/Makefile

@@ -7,6 +7,7 @@ include ../../../config.mak
 TARGET = ../libmltnormalize$(LIBSUF)
 
 OBJS = factory.o \
+	   filter_audiolevel.o \
 	   filter_volume.o 
 
 SRCS := $(OBJS:.o=.c)

src/modules/normalize/factory.c

@@ -22,6 +22,7 @@
 #include <limits.h>
 #include <framework/mlt.h>
 
+extern mlt_filter filter_audiolevel_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
 extern mlt_filter filter_volume_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg );
 
 static mlt_properties metadata( mlt_service_type type, const char *id, void *data )
@@ -33,6 +34,8 @@ static mlt_properties metadata( mlt_service_type type, const char *id, void *dat
 
 MLT_REPOSITORY
 {
+	MLT_REGISTER( filter_type, "audiolevel", filter_audiolevel_init );
 	MLT_REGISTER( filter_type, "volume", filter_volume_init );
+	MLT_REGISTER_METADATA( filter_type, "audiolevel", metadata, "filter_audiolevel.yml" );
 	MLT_REGISTER_METADATA( filter_type, "volume", metadata, "filter_volume.yml" );
 }

src/modules/normalize/filter_audiolevel.c

@@ -0,0 +1,131 @@
+/*
+ * filter_audiolevel.c -- get the audio level of each channel
+ * Copyright (C) 2002 Steve Harris
+ * Copyright (C) 2010 Marco Gittler <g.marco@freenet.de>
+ * Copyright (C) 2012 Dan Dennedy <dan@dennedy.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#include <framework/mlt_filter.h>
+#include <framework/mlt_frame.h>
+#include <framework/mlt_log.h>
+
+#include <stdlib.h>
+#include <math.h>
+
+#define AMPTODBFS(n) (log10(n) * 20.0)
+
+//----------------------------------------------------------------------------
+// IEC standard dB scaling -- as borrowed from meterbridge (c) Steve Harris
+
+static inline double IEC_Scale(double dB)
+{
+	double fScale = 1.0f;
+
+	if (dB < -70.0f)
+		fScale = 0.0f;
+	else if (dB < -60.0f)                 //  0.0  ..   2.5
+		fScale = (dB + 70.0f) * 0.0025f;
+	else if (dB < -50.0f)                 //  2.5  ..   7.5
+		fScale = (dB + 60.0f) * 0.005f + 0.025f;
+	else if (dB < -40.0)                  //  7.5  ..  15.0
+		fScale = (dB + 50.0f) * 0.0075f + 0.075f;
+	else if (dB < -30.0f)                 // 15.0  ..  30.0
+		fScale = (dB + 40.0f) * 0.015f + 0.15f;
+	else if (dB < -20.0f)                 // 30.0  ..  50.0
+		fScale = (dB + 30.0f) * 0.02f + 0.3f;
+	else if (dB < -0.001f || dB > 0.001f) // 50.0  .. 100.0
+		fScale = (dB + 20.0f) * 0.025f + 0.5f;
+
+	return fScale;
+}
+
+static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+{
+	mlt_filter filter = mlt_frame_pop_audio( frame );
+	int iec_scale = mlt_properties_get_int( MLT_FILTER_PROPERTIES(filter), "iec_scale" );
+	*format = mlt_audio_s16;
+	int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+	if ( error || !buffer ) return error;
+
+	int num_channels = *channels;
+	int num_samples = *samples > 200 ? 200 : *samples;
+	int num_oversample = 0;
+	int c, s;
+	char key[ 50 ];
+	int16_t *pcm = (int16_t*) *buffer;
+
+	for ( c = 0; c < *channels; c++ )
+	{
+		long val = 0;
+		double level = 0.0;
+
+		for ( s = 0; s < num_samples; s++ )
+		{
+			int sample = abs( pcm[c + s * num_channels] / 128 );
+			val += sample;
+			if ( sample == 128 )
+				num_oversample++;
+			else
+				num_oversample = 0;
+			// 10 samples @max => show max signal
+			if ( num_oversample > 10 )
+			{
+				level = 1.0;
+				break;
+			}
+			// if 3 samples over max => 1 peak over 0 db (0 dB = 40.0)
+			if ( num_oversample > 3 )
+				level = 41.0/42.0;
+		}
+		// max amplitude = 40/42, 3to10  oversamples=41, more then 10 oversamples=42
+		if ( level == 0.0 )
+			level = val / num_samples * 40.0/42.0 / 127.0;
+		if ( iec_scale )
+			level = IEC_Scale( AMPTODBFS( level ) );
+		sprintf( key, "meta.media.audio_level.%d", c );
+		mlt_properties_set_double( MLT_FRAME_PROPERTIES( frame ), key, level );
+		sprintf( key, "_audio_level.%d", c );
+		mlt_properties_set_double( MLT_FILTER_PROPERTIES( filter ), key, level );
+		mlt_log_debug( MLT_FILTER_SERVICE( filter ), "channel %d level %f\n", c, level );
+	}
+
+	return error;
+}
+
+/** Filter processing.
+*/
+
+static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
+{
+	mlt_frame_push_audio( frame, filter );
+	mlt_frame_push_audio( frame, filter_get_audio );
+	return frame;
+}
+
+/** Constructor for the filter.
+*/
+
+mlt_filter filter_audiolevel_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
+{
+	mlt_filter filter = mlt_filter_new();
+	if ( filter )
+	{
+		filter->process = filter_process;
+		mlt_properties_set_int( MLT_FILTER_PROPERTIES(filter), "iec_scale", 1 );
+	}
+	return filter;
+}

src/modules/normalize/filter_audiolevel.yml

@@ -0,0 +1,38 @@
+schema_version: 0.1
+type: filter
+identifier: audiolevel
+title: Audio Levels
+version: 1
+copyright: Dan Dennedy, Marco Gittler, and Steve Harris
+creator: Dan Dennedy
+contributor:
+  - Marco Gittler
+  - Steve Harris
+license: GPLv2
+language: en
+description: Compute the audio amplitude.
+notes: >
+  This filter provides the amplitude level as a percentage value in floating point.
+  This does not do any "slowing" of the data by averaging out peaks and
+  troughs of short duration like a VU meter.
+  Applications can also get this data on the frame as meta.media.audio_level.<N>
+  where <N> is the channel number starting with 0.
+tags:
+  - Audio
+parameters:
+  - identifier: iec_scale
+    title: Use IEC 60268-18 Scale
+    type: integer
+    minimum: 0
+    maximum: 1
+    default: 1
+    widget: checkbox
+
+  - identifier: _audio_level.<N>
+    description: >
+        <N> is the channel number starting with 0.
+        This is updated on every frame with audio.
+    readonly: yes
+    type: float
+    minimum: 0
+    maximum: 1