]> git.sesse.net Git - mlt/blobdiff - src/modules/sox/filter_sox.c
Merge pull request #25 from RedDwarf69/versioned_symbols
[mlt] / src / modules / sox / filter_sox.c
index 6cd8ae950d70da84f7a39513311c2d38978ae863..004c2be8a83c21818044e91599dc0b3e4b4f884a 100644 (file)
@@ -3,32 +3,56 @@
  * 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_filter.h>
 #include <framework/mlt_frame.h>
-#include "valerie/valerie_tokeniser.c"
+#include <framework/mlt_tokeniser.h>
+#include <framework/mlt_log.h>
 
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #include <math.h>
 
-#include <st.h>
+// TODO: does not support multiple effects with SoX v14.1.0+
+
+#ifdef SOX14
+#      include <sox.h>
+#      define ST_EOF SOX_EOF
+#      define ST_SUCCESS SOX_SUCCESS
+#      define st_sample_t sox_sample_t
+#      define eff_t sox_effect_t*
+#      define ST_LIB_VERSION_CODE SOX_LIB_VERSION_CODE
+#      define ST_LIB_VERSION SOX_LIB_VERSION
+#      if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,2,0))
+#              define st_size_t size_t
+#      else
+#              define st_size_t sox_size_t
+#      endif
+#      define ST_SIGNED_WORD_TO_SAMPLE(d,clips) SOX_SIGNED_16BIT_TO_SAMPLE(d,clips)
+#      if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
+#              define ST_SSIZE_MIN SOX_SAMPLE_MIN
+#      else
+#              define ST_SSIZE_MIN SOX_SSIZE_MIN
+#      endif
+#              define ST_SAMPLE_TO_SIGNED_WORD(d,clips) SOX_SAMPLE_TO_SIGNED_16BIT(d,clips)
+#else
+#      include <st.h>
+#endif
 
 #define BUFFER_LEN 8192
 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
@@ -56,50 +80,106 @@ static inline double mean( double *buf, int count )
        return mean;
 }
 
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
+static void delete_effect( eff_t effp )
+{
+       free( effp->priv );
+       free( (void*)effp->in_encoding );
+       free( effp );
+}
+#endif
+
 /** Create an effect state instance for a channels
 */
 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
 {
-       valerie_tokeniser tokeniser = valerie_tokeniser_init();
-       eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
+       mlt_tokeniser tokeniser = mlt_tokeniser_init();
        char id[ 256 ];
        int error = 1;
 
        // Tokenise the effect specification
-       valerie_tokeniser_parse_new( tokeniser, value, " " );
+       mlt_tokeniser_parse_new( tokeniser, value, " " );
+       if ( tokeniser->count < 1 )
+       {
+               mlt_tokeniser_close( tokeniser );
+               return error;
+       }
 
        // Locate the effect
+       mlt_destructor effect_destructor = mlt_pool_release;
+#ifdef SOX14
+       //fprintf(stderr, "%s: effect %s count %d\n", __FUNCTION__, tokeniser->tokens[0], tokeniser->count );
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
+       sox_effect_handler_t const *eff_handle = sox_find_effect( tokeniser->tokens[0] );
+       if (eff_handle == NULL ) return error;
+       eff_t eff = sox_create_effect( eff_handle );
+       effect_destructor = ( mlt_destructor ) delete_effect;
+       sox_encodinginfo_t *enc = calloc( 1, sizeof( sox_encodinginfo_t ) );
+       enc->encoding = SOX_ENCODING_SIGN2;
+       enc->bits_per_sample = 16;
+       eff->in_encoding = eff->out_encoding = enc;
+#else
+       eff_t eff = mlt_pool_alloc( sizeof( sox_effect_t ) );
+       sox_create_effect( eff, sox_find_effect( tokeniser->tokens[0] ) );
+#endif
+       int opt_count = tokeniser->count - 1;
+#else
+       eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
        int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
+#endif
        
        // If valid effect
        if ( opt_count != ST_EOF )
        {
                // Supply the effect parameters
+#ifdef SOX14
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,2,0))
+               if ( sox_effect_options( eff, opt_count, &tokeniser->tokens[ tokeniser->count > 1 ? 1 : 0  ] ) == ST_SUCCESS )
+#else
+               if ( ( * eff->handler.getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count > 1 ? 1 : 0  ] ) == ST_SUCCESS )
+#endif
+#else
                if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
+#endif
                {
                        // Set the sox signal parameters
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
+                       eff->in_signal.rate = frequency;
+                       eff->out_signal.rate = frequency;
+                       eff->in_signal.channels = 1;
+                       eff->out_signal.channels = 1;
+                       eff->in_signal.precision = 16;
+                       eff->out_signal.precision = 16;
+                       eff->in_signal.length = 0;
+                       eff->out_signal.length = 0;
+#else
                        eff->ininfo.rate = frequency;
                        eff->outinfo.rate = frequency;
                        eff->ininfo.channels = 1;
                        eff->outinfo.channels = 1;
+#endif
                        
                        // Start the effect
+#ifdef SOX14
+                       if ( ( * eff->handler.start )( eff ) == ST_SUCCESS )
+#else
                        if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
+#endif
                        {
                                // Construct id
                                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, effect_destructor, NULL );
                                error = 0;
                        }
                }
        }
        // Some error occurred so delete the temp effect state
        if ( error == 1 )
-               mlt_pool_release( eff );
+               effect_destructor( eff );
        
-       valerie_tokeniser_close( tokeniser );
+       mlt_tokeniser_close( tokeniser );
        
        return error;
 }
@@ -107,70 +187,29 @@ static int create_effect( mlt_filter this, char *value, int count, int channel,
 /** Get the audio.
 */
 
-static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
+static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
 {
-       // Get the properties of the frame
-       mlt_properties properties = mlt_frame_properties( frame );
-
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,3,0))
+       SOX_SAMPLE_LOCALS;
+#endif
        // 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 );
+
+       mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
 
        // Get the properties
-       st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
+       st_sample_t *input_buffer;// = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
        st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
-       int channels_avail = *channels;
        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 );
+       int count = mlt_properties_get_int( filter_properties, "_effect_count" );
+       int analysis = mlt_properties_get( filter_properties, "effect" ) && !strcmp( mlt_properties_get( filter_properties, "effect" ), "analysis" );
 
        // Get the producer's audio
-       mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
-
-       // Duplicate channels as necessary
-       if ( channels_avail < *channels )
-       {
-               int size = *channels * *samples * sizeof( int16_t );
-               int16_t *new_buffer = mlt_pool_alloc( size );
-               int j, k = 0;
-               
-               // Duplicate the existing channels
-               for ( i = 0; i < *samples; i++ )
-               {
-                       for ( j = 0; j < *channels; j++ )
-                       {
-                               new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
-                               k = ( k + 1 ) % channels_avail;
-                       }
-               }
-               
-               // Update the audio buffer now - destroys the old
-               mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-               
-               *buffer = new_buffer;
-       }
-       else if ( channels_avail == 6 && *channels == 2 )
-       {
-               // Nasty hack for ac3 5.1 audio - may be a cause of failure?
-               int size = *channels * *samples * sizeof( int16_t );
-               int16_t *new_buffer = mlt_pool_alloc( size );
-               
-               // Drop all but the first *channels
-               for ( i = 0; i < *samples; i++ )
-               {
-                       new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
-                       new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
-               }
-
-               // Update the audio buffer now - destroys the old
-               mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-               
-               *buffer = new_buffer;
-       }
+       *format = mlt_audio_s32;
+       mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
 
        // Even though some effects are multi-channel aware, it is not reliable
        // We must maintain a separate effect state for each channel
@@ -183,8 +222,13 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
                
                // Validate the existing effect state
+#if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
+               if ( e != NULL && ( e->in_signal.rate != *frequency || 
+                                                       e->out_signal.rate != *frequency ) )
+#else
                if ( e != NULL && ( e->ininfo.rate != *frequency || 
                                                        e->outinfo.rate != *frequency ) )
+#endif
                        e = NULL;
                
                // (Re)Create the effect state
@@ -205,7 +249,7 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                if ( !strncmp( name, "effect", 6 ) )
                                {
                                        // Get the effect specification
-                                       char *value = mlt_properties_get( filter_properties, name );
+                                       char *value = mlt_properties_get_value( filter_properties, j );
        
                                        // Create an instance
                                        if ( create_effect( filter, value, count, i, *frequency ) == 0 )
@@ -214,46 +258,96 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                        }
                        
                        // Save the number of filters
-                       mlt_properties_set_int( filter_properties, "effect_count", count );
+                       mlt_properties_set_int( filter_properties, "_effect_count", count );
                        
                }
-               if ( *samples > 0 && count > 0 )
+               if ( *samples > 0 && ( count > 0 || analysis ) )
                {
+                       input_buffer = (st_sample_t*) *buffer + i * *samples;
                        st_sample_t *p = input_buffer;
-                       st_sample_t *end = p + *samples;
-                       int16_t *q = *buffer + i;
                        st_size_t isamp = *samples;
                        st_size_t osamp = *samples;
-                       double rms = 0;
-                       int j;
+                       int j = *samples + 1;
                        char *normalise = mlt_properties_get( filter_properties, "normalise" );
                        double normalised_gain = 1.0;
                        
-                       // Convert to sox encoding
-                       while( p != end )
+                       if ( analysis )
                        {
-                               *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
-                               
-                               // Compute rms amplitude while we are accessing each sample
-                               rms += ( double )*p * ( double )*p;
-                               
-                               p ++;
-                               q += *channels;
+                               // Run analysis to compute a gain level to normalize the audio across entire filter duration
+                               double max_power = mlt_properties_get_double( filter_properties, "_max_power" );
+                               double peak = mlt_properties_get_double( filter_properties, "_max_peak" );
+                               double use_peak = mlt_properties_get_int( filter_properties, "use_peak" );
+                               double power = 0;
+                               int n = *samples + 1;
+
+                               // Compute power level of samples in this channel of this frame
+                               while ( --n )
+                               {
+                                       double s = fabs( *p++ );
+                                       // Track peak
+                                       if ( s > peak )
+                                       {
+                                               peak = s;
+                                               mlt_properties_set_double( filter_properties, "_max_peak", peak );
+                                       }
+                                       power += s * s;
+                               }
+                               power /= *samples;
+                               // Track maximum power
+                               if ( power > max_power )
+                               {
+                                       max_power = power;
+                                       mlt_properties_set_double( filter_properties, "_max_power", max_power );
+                               }
+
+                               // Complete analysis the last channel of the last frame.
+                               if ( i + 1 == *channels && mlt_filter_get_position( filter, frame ) + 1
+                                        == mlt_filter_get_length2( filter, frame ) )
+                               {
+                                       double rms = sqrt( max_power / ST_SSIZE_MIN / ST_SSIZE_MIN );
+                                       char effect[32];
+
+                                       // Convert RMS or peak to gain
+                                       if ( use_peak )
+                                               normalised_gain = ST_SSIZE_MIN / -peak;
+                                       else
+                                               normalised_gain = AMPLITUDE_NORM / rms;
+
+                                       // Set properties for serialization
+                                       snprintf( effect, sizeof(effect), "vol %f", normalised_gain );
+                                       effect[31] = 0;
+                                       mlt_properties_set( filter_properties, "effect", effect );
+                                       mlt_properties_set( filter_properties, "analyze", NULL );
+
+                                       // Show output comparable to normalize --no-adjust --fractions
+                                       mlt_properties_set_double( filter_properties, "level", rms );
+                                       mlt_properties_set_double( filter_properties, "gain", normalised_gain );
+                                       mlt_properties_set_double( filter_properties, "peak", -peak / ST_SSIZE_MIN );
+                               }
+
+                               // restore some variables
+                               p = input_buffer;
                        }
-                       
-                       // Compute final rms amplitude
-                       rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
-                       
+
                        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 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
                                if ( window > 0 && smooth_buffer != NULL )
                                {
@@ -292,7 +386,11 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                        float saved_gain = 1.0;
                                        
                                        // XXX: hack to apply the normalised gain level to the vol effect
+#ifdef SOX14
+                                       if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
+#else
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
+#endif
                                        {
                                                float *f = ( float * )( e->priv );
                                                saved_gain = *f;
@@ -300,35 +398,35 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format
                                        }
                                        
                                        // Apply the effect
-                                       if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
+#ifdef SOX14
+                                       if ( ( * e->handler.flow )( e, input_buffer, output_buffer, &isamp, &osamp ) != ST_SUCCESS )
+#else
+                                       if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) != ST_SUCCESS )
+#endif
                                        {
-                                               // Swap input and output buffer pointers for subsequent effects
-                                               p = input_buffer;
-                                               input_buffer = output_buffer;
-                                               output_buffer = p;
+                                               mlt_log_warning( MLT_FILTER_SERVICE(filter), "effect processing failed\n" );
                                        }
                                        
                                        // XXX: hack to restore the original vol gain to prevent accumulation
+#ifdef SOX14
+                                       if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
+#else
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
+#endif
                                        {
                                                float *f = ( float * )( e->priv );
                                                *f = saved_gain;
                                        }
                                }
                        }
-                       
-                       // Convert back to signed 16bit
-                       p = input_buffer;
-                       q = *buffer + i;
-                       end = p + *samples;
-                       while ( p != end )
-                       {
-                               *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
-                               q += *channels;
-                       }
+
+                       // Write back
+                       memcpy( input_buffer, output_buffer, *samples * sizeof(st_sample_t) );
                }
        }
 
+       mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
+
        return 0;
 }
 
@@ -337,15 +435,14 @@ 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 )
        {
                // 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;
+               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 )
                {
@@ -364,42 +461,38 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 /** Constructor for the filter.
 */
 
-mlt_filter filter_sox_init( char *arg )
+mlt_filter filter_sox_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 {
        mlt_filter this = mlt_filter_new( );
        if ( this != NULL )
        {
                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;
                
-               if ( arg != NULL )
+               if ( !strncmp( id, "sox.", 4 ) )
+               {
+                       char *s = malloc( strlen( id ) + ( arg? strlen( arg ) + 2 : 1 ) );
+                       strcpy( s, id + 4 );
+                       if ( arg )
+                       {
+                               strcat( s, " " );
+                               strcat( s, arg );
+                       }
+                       mlt_properties_set( properties, "effect", s );
+                       free( s );
+               }
+               else if ( arg )
                        mlt_properties_set( properties, "effect", arg );
                mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
                mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
                mlt_properties_set_int( properties, "window", 75 );
+               mlt_properties_set( properties, "version", sox_version() );
        }
        return this;
 }
 
 // What to do when a libst internal failure occurs
 void cleanup(void){}
-
-// Is there a build problem with my sox-devel package?
-#ifndef gsm_create
-void gsm_create(void){}
-#endif
-#ifndef gsm_decode
-void gsm_decode(void){}
-#endif
-#ifndef gdm_encode
-void gsm_encode(void){}
-#endif
-#ifndef gsm_destroy
-void gsm_destroy(void){}
-#endif
-#ifndef gsm_option
-void gsm_option(void){}
-#endif