+++ /dev/null
-/*
- * filter_volume.c -- adjust audio volume
- * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
- * Author: 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 <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <ctype.h>
-#include <string.h>
-
-#define MAX_CHANNELS 6
-#define EPSILON 0.00001
-
-/* The following normalise functions come 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 = x;
-
- if (x < -lmtr_lvl)
- xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl;
- else if (x > lmtr_lvl)
- xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl;
-
-// if ( x != xp )
-// fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp );
-
- 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 );
-}
-
-/* ------ End normalize functions --------------------------------------- */
-
-/** Get the audio.
-*/
-
-static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
-{
- // Get the properties of the a frame
- mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
- double gain = mlt_properties_get_double( properties, "volume.gain" );
- double max_gain = mlt_properties_get_double( properties, "volume.max_gain" );
- double limiter_level = 0.5; /* -6 dBFS */
- int normalise = mlt_properties_get_int( properties, "volume.normalise" );
- double amplitude = mlt_properties_get_double( properties, "volume.amplitude" );
- int i, j;
- double sample;
- int16_t peak;
-
- // Get the filter from the frame
- mlt_filter this = mlt_properties_get_data( properties, "filter_volume", NULL );
-
- // Get the properties from the filter
- mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
-
- if ( mlt_properties_get( properties, "volume.limiter" ) != NULL )
- limiter_level = mlt_properties_get_double( properties, "volume.limiter" );
-
- // Get the producer's audio
- mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
-// fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
-
- // 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 ( normalise )
- {
- int window = mlt_properties_get_int( filter_props, "window" );
- double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
-
- if ( window > 0 && smooth_buffer != NULL )
- {
- int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
-
- // Compute the signal power and put into smoothing buffer
- smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
-// fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
- if ( smooth_buffer[ smooth_index ] > EPSILON )
- {
- mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
-
- // Smooth the data and compute the gain
-// fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
- gain *= amplitude / get_smoothed_data( smooth_buffer, window );
- }
- }
- else
- {
- gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak );
- }
- }
-
-// if ( gain > 1.0 && normalise )
-// fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
-
- if ( max_gain > 0 && gain > max_gain )
- gain = max_gain;
-
- // Initialise filter's previous gain value to prevent an inadvertant jump from 0
- mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
- mlt_position current_position = mlt_frame_get_position( frame );
- if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
- || current_position != last_position + 1 )
- mlt_properties_set_double( filter_props, "_previous_gain", gain );
-
- // Start the gain out at the previous
- double previous_gain = mlt_properties_get_double( filter_props, "_previous_gain" );
-
- // Determine ramp increment
- double gain_step = ( gain - previous_gain ) / *samples;
-// fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
-
- // Save the current gain for the next iteration
- mlt_properties_set_double( filter_props, "_previous_gain", gain );
- mlt_properties_set_position( filter_props, "_last_position", current_position );
-
- // Ramp from the previous gain to the current
- gain = previous_gain;
-
- int16_t *p = *buffer;
-
- // Apply the gain
- for ( i = 0; i < *samples; i++ )
- {
- for ( j = 0; j < *channels; j++ )
- {
- sample = *p * gain;
- *p = ROUND( sample );
-
- if ( gain > 1.0 )
- {
- /* use limiter function instead of clipping */
- if ( normalise )
- *p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
-
- /* perform clipping */
- else if ( sample > samplemax )
- *p = samplemax;
- else if ( sample < samplemin )
- *p = samplemin;
- }
- p++;
- }
- gain += gain_step;
- }
-
- return 0;
-}
-
-/** Filter processing.
-*/
-
-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 );
-
- // Parse the gain property
- if ( mlt_properties_get( properties, "gain" ) == NULL )
- {
- double gain = 1.0; // no adjustment
-
- if ( mlt_properties_get( filter_props, "gain" ) != NULL )
- {
- char *p = mlt_properties_get( filter_props, "gain" );
-
- if ( strncaseeq( p, "normalise", 9 ) )
- mlt_properties_set( filter_props, "normalise", "" );
- else
- {
- if ( strcmp( p, "" ) != 0 )
- gain = fabs( strtod( p, &p) );
-
- while ( isspace( *p ) )
- p++;
-
- /* check if "dB" is given after number */
- if ( strncaseeq( p, "db", 2 ) )
- gain = DBFSTOAMP( gain );
-
- // If there is an end adjust gain to the range
- if ( mlt_properties_get( filter_props, "end" ) != NULL )
- {
- // Determine the time position of this frame in the transition duration
- mlt_position in = mlt_filter_get_in( this );
- mlt_position out = mlt_filter_get_out( this );
- mlt_position time = mlt_frame_get_position( frame );
- double position = ( double )( time - in ) / ( double )( out - in + 1 );
-
- double end = -1;
- char *p = mlt_properties_get( filter_props, "end" );
- if ( strcmp( p, "" ) != 0 )
- end = fabs( strtod( p, &p) );
-
- while ( isspace( *p ) )
- p++;
-
- /* check if "dB" is given after number */
- if ( strncaseeq( p, "db", 2 ) )
- end = DBFSTOAMP( gain );
-
- if ( end != -1 )
- gain += ( end - gain ) * position;
- }
- }
- }
- mlt_properties_set_double( properties, "volume.gain", gain );
- }
-
- // Parse the maximum gain property
- if ( mlt_properties_get( filter_props, "max_gain" ) != NULL )
- {
- char *p = mlt_properties_get( filter_props, "max_gain" );
- double gain = fabs( strtod( p, &p) ); // 0 = no max
-
- while ( isspace( *p ) )
- p++;
-
- /* check if "dB" is given after number */
- if ( strncaseeq( p, "db", 2 ) )
- gain = DBFSTOAMP( gain );
-
- mlt_properties_set_double( properties, "volume.max_gain", gain );
- }
-
- // Parse 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);
-
- while ( isspace( *p ) )
- p++;
-
- /* check if "dB" is given after number */
- if ( strncaseeq( p, "db", 2 ) )
- {
- if ( level > 0 )
- level = -level;
- level = DBFSTOAMP( level );
- }
- else
- {
- if ( level < 0 )
- level = -level;
- }
- mlt_properties_set_double( properties, "volume.limiter", level );
- }
-
- // Parse 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);
-
- while ( isspace( *p ) )
- p++;
-
- /* check if "dB" is given after number */
- 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;
- }
-
- // If there is an end adjust gain to the range
- if ( mlt_properties_get( filter_props, "end" ) != NULL )
- {
- // Determine the time position of this frame in the transition duration
- mlt_position in = mlt_filter_get_in( this );
- mlt_position out = mlt_filter_get_out( this );
- mlt_position time = mlt_frame_get_position( frame );
- double position = ( double )( time - in ) / ( double )( out - in + 1 );
- amplitude *= position;
- }
- mlt_properties_set_int( properties, "volume.normalise", 1 );
- mlt_properties_set_double( properties, "volume.amplitude", amplitude );
- }
-
- // Parse the window property and allocate smoothing buffer if needed
- int window = mlt_properties_get_int( filter_props, "window" );
- if ( mlt_properties_get( filter_props, "smooth_buffer" ) == NULL && window > 1 )
- {
- // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
- double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
- int i;
- for ( i = 0; i < window; i++ )
- smooth_buffer[ i ] = -1.0;
- mlt_properties_set_data( filter_props, "smooth_buffer", smooth_buffer, 0, free, NULL );
- }
-
- // Put a filter reference onto the frame
- mlt_properties_set_data( properties, "filter_volume", this, 0, NULL, NULL );
-
- // Override the get_audio method
- mlt_frame_push_audio( frame, filter_get_audio );
-
- return frame;
-}
-
-/** Constructor for the filter.
-*/
-
-mlt_filter filter_volume_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
-{
- mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 );
- if ( this != NULL && mlt_filter_init( this, NULL ) == 0 )
- {
- mlt_properties properties = MLT_FILTER_PROPERTIES( this );
- this->process = filter_process;
- if ( arg != NULL )
- mlt_properties_set( properties, "gain", arg );
-
- mlt_properties_set_int( properties, "window", 75 );
- mlt_properties_set( properties, "max_gain", "20dB" );
- }
- return this;
-}
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