#include <stdio.h>
#include <stdlib.h>
+#include <math.h>
+#include <ctype.h>
+#include <string.h>
+
+#define MAX_CHANNELS 6
+#define SMOOTH_BUFFER_SIZE 50
+
+/* This utilities and limiter function comes from the normalize utility:
+ Copyright (C) 1999--2002 Chris Vaill */
+
+#define samp_width 16
+
+#ifndef ROUND
+# define ROUND(x) floor((x) + 0.5)
+#endif
+
+#define DBFSTOAMP(x) pow(10,(x)/20.0)
+
+/** Return nonzero if the two strings are equal, ignoring case, up to
+ the first n characters.
+*/
+int strncaseeq(const char *s1, const char *s2, size_t n)
+{
+ for ( ; n > 0; n--)
+ {
+ if (tolower(*s1++) != tolower(*s2++))
+ return 0;
+ }
+ return 1;
+}
+
+/** Limiter function.
+
+ / tanh((x + lev) / (1-lev)) * (1-lev) - lev (for x < -lev)
+ |
+ x' = | x (for |x| <= lev)
+ |
+ \ tanh((x - lev) / (1-lev)) * (1-lev) + lev (for x > lev)
+
+ With limiter level = 0, this is equivalent to a tanh() function;
+ with limiter level = 1, this is equivalent to clipping.
+*/
+static inline double limiter( double x, double lmtr_lvl )
+{
+ double xp;
+
+ if (x < -lmtr_lvl)
+ xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl;
+ else if (x <= lmtr_lvl)
+ xp = x;
+ else
+ xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl;
+
+ return xp;
+}
+
+
+/** Takes a full smoothing window, and returns the value of the center
+ element, smoothed.
+
+ Currently, just does a mean filter, but we could do a median or
+ gaussian filter here instead.
+*/
+static inline double get_smoothed_data( double *buf, int count )
+{
+ int i, j;
+ double smoothed = 0;
+
+ for ( i = 0, j = 0; i < count; i++ )
+ {
+ if ( buf[ i ] != -1.0 )
+ {
+ smoothed += buf[ i ];
+ j++;
+ }
+ }
+ smoothed /= j;
+// fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed );
+
+ return smoothed;
+}
+
+/** Get the max power level (using RMS) and peak level of the audio segment.
+ */
+double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *peak )
+{
+ // Determine numeric limits
+ int bytes_per_samp = (samp_width - 1) / 8 + 1;
+ int16_t max = (1 << (bytes_per_samp * 8 - 1)) - 1;
+ int16_t min = -max - 1;
+
+ double *sums = (double *) calloc( channels, sizeof(double) );
+ int c, i;
+ int16_t sample;
+ double pow, maxpow = 0;
+
+ /* initialize peaks to effectively -inf and +inf */
+ int16_t max_sample = min;
+ int16_t min_sample = max;
+
+ for ( i = 0; i < samples; i++ )
+ {
+ for ( c = 0; c < channels; c++ )
+ {
+ sample = *buffer++;
+ sums[ c ] += (double) sample * (double) sample;
+
+ /* track peak */
+ if ( sample > max_sample )
+ max_sample = sample;
+ else if ( sample < min_sample )
+ min_sample = sample;
+ }
+ }
+ for ( c = 0; c < channels; c++ )
+ {
+ pow = sums[ c ] / (double) samples;
+ if ( pow > maxpow )
+ maxpow = pow;
+ }
+
+ free( sums );
+
+ /* scale the pow value to be in the range 0.0 -- 1.0 */
+ maxpow /= ( (double) min * (double) min);
+
+ if ( -min_sample > max_sample )
+ *peak = min_sample / (double) min;
+ else
+ *peak = max_sample / (double) max;
+
+ return sqrt( maxpow );
+}
/** Get the audio.
*/
{
// Get the properties of the a frame
mlt_properties properties = mlt_frame_properties( frame );
- double volume = mlt_properties_get_double( properties, "volume" );
+ double gain = mlt_properties_get_double( properties, "gain" );
+ int use_limiter = mlt_properties_get_int( properties, "volume.use_limiter" );
+ double limiter_level = mlt_properties_get_double( properties, "volume.limiter_level" );
+ int normalise = mlt_properties_get_int( properties, "volume.normalise" );
+ double amplitude = mlt_properties_get_double( properties, "volume.amplitude" );
+ int i;
+ double sample;
+ int16_t peak;
// Restore the original get_audio
frame->get_audio = mlt_properties_get_data( properties, "volume.get_audio", NULL );
// Get the producer's audio
mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
- // Apply the volume
- int i;
+ // Determine numeric limits
+ int bytes_per_samp = (samp_width - 1) / 8 + 1;
+ int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
+ int samplemin = -samplemax - 1;
+
+#if 0
+ if ( gain > 1.0 && use_limiter != 0 )
+ fprintf(stderr, "filter_volume: limiting samples greater than %f\n", limiter_level );
+#endif
+
+ if ( normalise )
+ {
+ double *smooth_buffer = mlt_properties_get_data( properties, "volume.smooth_buffer", NULL );
+ int *smooth_index = mlt_properties_get_data( properties, "volume.smooth_index", NULL );
+
+ // Compute the signal power and put into smoothing buffer
+ smooth_buffer[ *smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
+ *smooth_index = ( *smooth_index + 1 ) % SMOOTH_BUFFER_SIZE;
+
+ // Smooth the data and compute the gain
+ gain *= amplitude / get_smoothed_data( smooth_buffer, SMOOTH_BUFFER_SIZE );
+ }
+
+ // Apply the gain
for ( i = 0; i < ( *channels * *samples ); i++ )
- (*buffer)[i] *= volume;
+ {
+ sample = (*buffer)[i] * gain;
+ (*buffer)[i] = ROUND( sample );
+
+ if ( gain > 1.0 )
+ {
+ /* use limiter function instead of clipping */
+ if ( use_limiter != 0 )
+ (*buffer)[i] = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
+
+ /* perform clipping */
+ else if ( sample > samplemax )
+ (*buffer)[i] = samplemax;
+ else if ( sample < samplemin )
+ (*buffer)[i] = samplemin;
+ }
+ }
return 0;
}
static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
{
mlt_properties properties = mlt_frame_properties( frame );
+ mlt_properties filter_props = mlt_filter_properties( this );
- // Propogate the level property
- if ( mlt_properties_get( mlt_filter_properties( this ), "volume" ) != NULL )
- mlt_properties_set_double( properties, "volume",
- mlt_properties_get_double( mlt_filter_properties( this ), "volume" ) );
+ // Propogate the volume/gain property
+ if ( mlt_properties_get( properties, "gain" ) == NULL )
+ {
+ double gain = 1.0; // none
+ if ( mlt_properties_get( filter_props, "volume" ) != NULL )
+ gain = mlt_properties_get_double( filter_props, "volume" );
+ if ( mlt_properties_get( filter_props, "gain" ) != NULL )
+ gain = mlt_properties_get_double( filter_props, "gain" );
+ mlt_properties_set_double( properties, "gain", gain );
+ }
+ // Parse and propogate the limiter property
+ if ( mlt_properties_get( filter_props, "limiter" ) != NULL )
+ {
+ char *p = mlt_properties_get( filter_props, "limiter" );
+ double level = 0.5; /* -6dBFS */
+ if ( strcmp( p, "" ) != 0 )
+ level = strtod( p, &p);
+
+ /* check if "dB" is given after number */
+ while ( isspace( *p ) )
+ p++;
+
+ if ( strncaseeq( p, "db", 2 ) )
+ {
+ if ( level > 0 )
+ level = -level;
+ level = DBFSTOAMP( level );
+ }
+ else
+ {
+ if ( level < 0 )
+ level = -level;
+ }
+ mlt_properties_set_int( properties, "volume.use_limiter", 1 );
+ mlt_properties_set_double( properties, "volume.limiter_level", level );
+ }
+
+ // Parse and propogate the normalise property
+ if ( mlt_properties_get( filter_props, "normalise" ) != NULL )
+ {
+ char *p = mlt_properties_get( filter_props, "normalise" );
+ double amplitude = 0.2511886431509580; /* -12dBFS */
+ if ( strcmp( p, "" ) != 0 )
+ amplitude = strtod( p, &p);
+
+ /* check if "dB" is given after number */
+ while ( isspace( *p ) )
+ p++;
+
+ if ( strncaseeq( p, "db", 2 ) )
+ {
+ if ( amplitude > 0 )
+ amplitude = -amplitude;
+ amplitude = DBFSTOAMP( amplitude );
+ }
+ else
+ {
+ if ( amplitude < 0 )
+ amplitude = -amplitude;
+ if ( amplitude > 1.0 )
+ amplitude = 1.0;
+ }
+ mlt_properties_set_int( properties, "volume.normalise", 1 );
+ mlt_properties_set_double( properties, "volume.amplitude", amplitude );
+ }
+
+ // Propogate the smoothing buffer properties
+ mlt_properties_set_data( properties, "volume.smooth_buffer",
+ mlt_properties_get_data( filter_props, "smooth_buffer", NULL ), 0, NULL, NULL );
+ mlt_properties_set_data( properties, "volume.smooth_index",
+ mlt_properties_get_data( filter_props, "smooth_index", NULL ), 0, NULL, NULL );
+
// Backup the original get_audio (it's still needed)
mlt_properties_set_data( properties, "volume.get_audio", frame->get_audio, 0, NULL, NULL );
this->process = filter_process;
if ( arg != NULL )
mlt_properties_set_double( mlt_filter_properties( this ), "volume", atof( arg ) );
+
+ // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
+ double *smooth_buffer = (double*) calloc( SMOOTH_BUFFER_SIZE, sizeof( double ) );
+ int i;
+ for ( i = 0; i < SMOOTH_BUFFER_SIZE; i++ )
+ smooth_buffer[ i ] = -1.0;
+ mlt_properties_set_data( mlt_filter_properties( this ), "smooth_buffer", smooth_buffer, 0, free, NULL );
+ int *smooth_index = calloc( 1, sizeof( int ) );
+ mlt_properties_set_data( mlt_filter_properties( this ), "smooth_index", smooth_index, 0, free, NULL );
}
return this;
}
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