Only compute rms if normalise.

diff --git a/src/modules/sox/filter_sox.c b/src/modules/sox/filter_sox.c
index cc73d1f01655768a72a8a7c6608b7b265147b154..32860586b26b0e11f19400f3bdd2a91eafd272ee 100644 (file)
--- a/src/modules/sox/filter_sox.c
+++ b/src/modules/sox/filter_sox.c
@@ -263,30 +263,27 @@ static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *f
                        st_sample_t *p = input_buffer;
                        st_size_t isamp = *samples;
                        st_size_t osamp = *samples;
-                       double rms = 0;
                        int j = *samples + 1;
                        char *normalise = mlt_properties_get( filter_properties, "normalise" );
                        double normalised_gain = 1.0;
                        
-                       // Convert from interleaved
-                       while( --j )
-                       {
-                               // Compute rms amplitude while we are accessing each sample
-                               rms += ( double )*p * ( double )*p;
-                               p ++;
-                       }
-                       
                        if ( normalise )
                        {
                                int window = mlt_properties_get_int( filter_properties, "window" );
                                double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
                                double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
-                               
+                               double rms = 0;
+
                                // Default the maximum gain factor to 20dBFS
                                if ( max_gain == 0 )
                                        max_gain = 10.0;
                                
-                               // Compute final rms amplitude
+                               // Compute rms amplitude
+                               while( --j )
+                               {
+                                       rms += ( double )*p * ( double )*p;
+                                       p ++;
+                               }
                                rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
 
                                // The smoothing buffer prevents radical shifts in the gain level