add sox 13.0.0 support
[melted] / src / modules / sox / filter_sox.c
index 5056320..7c0928b 100644 (file)
@@ -3,25 +3,25 @@
  * 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 library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful,
+ * This library 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.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser 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.
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
 
 #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 );
@@ -90,7 +90,7 @@ static int create_effect( mlt_filter this, char *value, int count, int channel,
                                sprintf( id, "_effect_%d_%d", count, channel );
 
                                // Save the effect state
-                               mlt_properties_set_data( mlt_filter_properties( this ), id, eff, 0, mlt_pool_release, NULL );
+                               mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, mlt_pool_release, NULL );
                                error = 0;
                        }
                }
@@ -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;
 }
@@ -110,13 +110,13 @@ static int create_effect( mlt_filter this, char *value, int count, int channel,
 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 frame
-       mlt_properties properties = mlt_frame_properties( frame );
+       mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 
        // Get the filter service
        mlt_filter filter = mlt_frame_pop_audio( frame );
 
        // Get the filter properties
-       mlt_properties filter_properties = mlt_filter_properties( filter );
+       mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
 
        // Get the properties
        st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
@@ -125,9 +125,6 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
        int i; // channel
        int count = mlt_properties_get_int( filter_properties, "effect_count" );
 
-       // Restore the original get_audio
-       frame->get_audio = mlt_frame_pop_audio( frame );
-
        // Get the producer's audio
        mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
 
@@ -228,12 +225,18 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                        int j;
                        char *normalise = mlt_properties_get( filter_properties, "normalise" );
                        double normalised_gain = 1.0;
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
+                       st_sample_t dummy_clipped_count = 0;
+#endif
                        
                        // Convert to sox encoding
                        while( p != end )
                        {
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
+                               *p = ST_SIGNED_WORD_TO_SAMPLE( *q, dummy_clipped_count );
+#else
                                *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
-                               
+#endif
                                // Compute rms amplitude while we are accessing each sample
                                rms += ( double )*p * ( double )*p;
                                
@@ -250,15 +253,21 @@ 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 +277,8 @@ 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 +289,12 @@ 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 +302,16 @@ 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 );
@@ -311,7 +326,11 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                        end = p + *samples;
                        while ( p != end )
                        {
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
+                               *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++, dummy_clipped_count );
+#else
                                *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
+#endif
                                q += *channels;
                        }
                }
@@ -325,14 +344,14 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
 
 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 {
-       if ( frame->get_audio != NULL )
+       if ( mlt_frame_is_test_audio( frame ) == 0 )
        {
-               mlt_frame_push_audio( frame, frame->get_audio );
+               // Add the filter to the frame
                mlt_frame_push_audio( frame, this );
-               frame->get_audio = filter_get_audio;
+               mlt_frame_push_audio( frame, filter_get_audio );
                
                // Parse the window property and allocate smoothing buffer if needed
-               mlt_properties properties = mlt_filter_properties( this );
+               mlt_properties properties = MLT_FILTER_PROPERTIES( this );
                int window = mlt_properties_get_int( properties, "window" );
                if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
                {
@@ -358,7 +377,7 @@ mlt_filter filter_sox_init( char *arg )
        {
                void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
                void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
-               mlt_properties properties = mlt_filter_properties( this );
+               mlt_properties properties = MLT_FILTER_PROPERTIES( this );
                
                this->process = filter_process;