Rectangle added to properties

[melted] / src / modules / sox / filter_sox.c

diff --git a/src/modules/sox/filter_sox.c b/src/modules/sox/filter_sox.c
index 5056320..b707f5b 100644 (file)
--- a/src/modules/sox/filter_sox.c
+++ b/src/modules/sox/filter_sox.c
@@ -21,7 +21,7 @@
 #include "filter_sox.h"
 
 #include <framework/mlt_frame.h>
-#include "valerie/valerie_tokeniser.c"
+#include <framework/mlt_tokeniser.h>
 
 #include <stdio.h>
 #include <stdlib.h>
@@ -60,13 +60,13 @@ static inline double mean( double *buf, int count )
 */
 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
 {
-       valerie_tokeniser tokeniser = valerie_tokeniser_init();
+       mlt_tokeniser tokeniser = mlt_tokeniser_init();
        eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
        char id[ 256 ];
        int error = 1;
 
        // Tokenise the effect specification
-       valerie_tokeniser_parse_new( tokeniser, value, " " );
+       mlt_tokeniser_parse_new( tokeniser, value, " " );
 
        // Locate the effect
        int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
@@ -99,7 +99,7 @@ static int create_effect( mlt_filter this, char *value, int count, int channel,
        if ( error == 1 )
                mlt_pool_release( eff );
        
-       valerie_tokeniser_close( tokeniser );
+       mlt_tokeniser_close( tokeniser );
        
        return error;
 }
@@ -250,15 +250,21 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
                                double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
                                
+                               // Default the maximum gain factor to 20dBFS
                                if ( max_gain == 0 )
                                        max_gain = 10.0;
                                
+                               // The smoothing buffer prevents radical shifts in the gain level
                                if ( window > 0 && smooth_buffer != NULL )
                                {
                                        int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
                                        smooth_buffer[ smooth_index ] = rms;
+                                       
+                                       // Ignore very small values that adversely affect the mean
                                        if ( rms > AMPLITUDE_MIN )
                                                mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
+                                       
+                                       // Smoothing is really just a mean over the past N values
                                        normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
                                }
                                else if ( rms > 0 )
@@ -268,6 +274,8 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                }
                                        
                                //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
+                               
+                               // Govern the maximum gain
                                if ( normalised_gain > max_gain )
                                        normalised_gain = max_gain;
                        }
@@ -278,11 +286,12 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                sprintf( id, "_effect_%d_%d", j, i );
                                e = mlt_properties_get_data( filter_properties, id, NULL );
                                
-                               // Apply the effect
+                               // We better have this guy
                                if ( e != NULL )
                                {
                                        float saved_gain = 1.0;
                                        
+                                       // XXX: hack to apply the normalised gain level to the vol effect
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
                                        {
                                                float *f = ( float * )( e->priv );
@@ -290,13 +299,16 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                                *f = saved_gain * normalised_gain;
                                        }
                                        
+                                       // Apply the effect
                                        if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
                                        {
+                                               // Swap input and output buffer pointers for subsequent effects
                                                p = input_buffer;
                                                input_buffer = output_buffer;
                                                output_buffer = p;
                                        }
                                        
+                                       // XXX: hack to restore the original vol gain to prevent accumulation
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
                                        {
                                                float *f = ( float * )( e->priv );
@@ -327,6 +339,7 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 {
        if ( frame->get_audio != NULL )
        {
+               // Add the filter to the frame
                mlt_frame_push_audio( frame, frame->get_audio );
                mlt_frame_push_audio( frame, this );
                frame->get_audio = filter_get_audio;