#include <string.h>
#include <math.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_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)
-# define ST_SSIZE_MIN SOX_SSIZE_MIN
-# 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
*/
static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
{
mlt_tokeniser tokeniser = mlt_tokeniser_init();
-#ifdef SOX14
- eff_t eff = mlt_pool_alloc( sizeof( sox_effect_t ) );
-#else
- eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
-#endif
char id[ 256 ];
int error = 1;
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
#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
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 );
mlt_tokeniser_close( tokeniser );
/** 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 );
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 );
+ 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" );
// 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
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
}
if ( *samples > 0 && count > 0 )
{
+ 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 )
+ // Convert from interleaved
+ while( --j )
{
-#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;
}
// Compute final rms amplitude
}
}
}
-
- // 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( output_buffer, input_buffer, *samples * sizeof(st_sample_t) );
}
}
projects / mlt / blobdiff