#include <framework/mlt_filter.h>
#include <framework/mlt_frame.h>
#include <framework/mlt_tokeniser.h>
+#include <framework/mlt_log.h>
#include <stdio.h>
#include <stdlib.h>
# define ST_SUCCESS SOX_SUCCESS
# define st_sample_t sox_sample_t
# define eff_t sox_effect_t*
-# define st_size_t sox_size_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)
+# 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
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
*/
// Tokenise the effect specification
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))
- eff_t eff = sox_create_effect( sox_find_effect( tokeniser->tokens[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;
{
// 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
/** 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_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" );
+ 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
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 )
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;
-#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 ( analysis )
{
-#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;
-
- 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 )
{
// Apply the effect
#ifdef SOX14
- if ( ( * e->handler.flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
+ 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 )
+ 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
}
}
}
-
- // Convert back to signed 16bit
- p = input_buffer;
- q = *buffer + i;
- 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;
- }
+
+ // Write back
+ memcpy( input_buffer, output_buffer, *samples * sizeof(st_sample_t) );
}
}
+ mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
+
return 0;
}
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
projects / mlt / blobdiff