static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the filter from the frame
mlt_filter this = mlt_frame_pop_audio( frame );
// Get the properties from the filter
mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
// Get the frame's filter instance properties
mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( this ) );
// Get the parameters
double gain = mlt_properties_get_double( instance_props, "gain" );
double max_gain = mlt_properties_get_double( instance_props, "max_gain" );
double limiter_level = 0.5; /* -6 dBFS */
int normalise = mlt_properties_get_int( instance_props, "normalise" );
double amplitude = mlt_properties_get_double( instance_props, "amplitude" );
int i, j;
double sample;
int16_t peak;
if ( mlt_properties_get( instance_props, "limiter" ) != NULL )
limiter_level = mlt_properties_get_double( instance_props, "limiter" );
// Get the producer's audio
*format = mlt_audio_s16;
mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
// fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
// Determine numeric limits
int bytes_per_samp = (samp_width - 1) / 8 + 1;
int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
int samplemin = -samplemax - 1;
mlt_service_lock( MLT_FILTER_SERVICE( this ) );
if ( normalise )
{
int window = mlt_properties_get_int( filter_props, "window" );
double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
if ( window > 0 && smooth_buffer != NULL )
{
int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
// Compute the signal power and put into smoothing buffer
smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
// fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
if ( smooth_buffer[ smooth_index ] > EPSILON )
{
mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
// Smooth the data and compute the gain
// fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
gain *= amplitude / get_smoothed_data( smooth_buffer, window );
}
}
else
{
gain *= amplitude / signal_max_power( (int16_t*) *buffer, *channels, *samples, &peak );
}
}
// if ( gain > 1.0 && normalise )
// fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
if ( max_gain > 0 && gain > max_gain )
gain = max_gain;
// Initialise filter's previous gain value to prevent an inadvertant jump from 0
mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
mlt_position current_position = mlt_frame_get_position( frame );
if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
|| current_position != last_position + 1 )
mlt_properties_set_double( filter_props, "_previous_gain", gain );
// Start the gain out at the previous
double previous_gain = mlt_properties_get_double( filter_props, "_previous_gain" );
// Determine ramp increment
double gain_step = ( gain - previous_gain ) / *samples;
// fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
// Save the current gain for the next iteration
mlt_properties_set_double( filter_props, "_previous_gain", gain );
mlt_properties_set_position( filter_props, "_last_position", current_position );
mlt_service_unlock( MLT_FILTER_SERVICE( this ) );
// Ramp from the previous gain to the current
gain = previous_gain;
int16_t *p = (int16_t*) *buffer;
// Apply the gain
for ( i = 0; i < *samples; i++ )
{
for ( j = 0; j < *channels; j++ )
{
sample = *p * gain;
*p = ROUND( sample );
if ( gain > 1.0 )
{
/* use limiter function instead of clipping */
if ( normalise )
*p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
/* perform clipping */
else if ( sample > samplemax )
*p = samplemax;
else if ( sample < samplemin )
*p = samplemin;
}
p++;
}
gain += gain_step;
}
return 0;
}
static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the filter from the frame
mlt_filter filter = mlt_frame_pop_audio( frame );
// Get the properties from the filter
mlt_properties filter_props = MLT_FILTER_PROPERTIES( filter );
// Get the frame's filter instance properties
mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( filter ) );
// Get the parameters
double gain = mlt_properties_get_double( instance_props, "gain" );
double max_gain = mlt_properties_get_double( instance_props, "max_gain" );
double limiter_level = 0.5; /* -6 dBFS */
int normalise = mlt_properties_get_int( instance_props, "normalise" );
double amplitude = mlt_properties_get_double( instance_props, "amplitude" );
int i, j;
double sample;
int16_t peak;
// Use animated value for gain if "level" property is set
char* level_property = mlt_properties_get( filter_props, "level" );
if ( level_property != NULL )
{
mlt_position position = mlt_filter_get_position( filter, frame );
mlt_position length = mlt_filter_get_length2( filter, frame );
gain = mlt_properties_anim_get_double( filter_props, "level", position, length );
gain = DBFSTOAMP( gain );
}
if ( mlt_properties_get( instance_props, "limiter" ) != NULL )
limiter_level = mlt_properties_get_double( instance_props, "limiter" );
// Get the producer's audio
*format = normalise? mlt_audio_s16 : mlt_audio_f32le;
mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
if ( normalise )
{
int window = mlt_properties_get_int( filter_props, "window" );
double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
if ( window > 0 && smooth_buffer != NULL )
{
int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
// Compute the signal power and put into smoothing buffer
smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
if ( smooth_buffer[ smooth_index ] > EPSILON )
{
mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
// Smooth the data and compute the gain
gain *= amplitude / get_smoothed_data( smooth_buffer, window );
}
}
else
{
gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak );
}
}
if ( max_gain > 0 && gain > max_gain )
gain = max_gain;
// Initialise filter's previous gain value to prevent an inadvertant jump from 0
mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
mlt_position current_position = mlt_frame_get_position( frame );
if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
|| current_position != last_position + 1 )
mlt_properties_set_double( filter_props, "_previous_gain", gain );
// Start the gain out at the previous
double previous_gain = mlt_properties_get_double( filter_props, "_previous_gain" );
// Determine ramp increment
double gain_step = ( gain - previous_gain ) / *samples;
// Save the current gain for the next iteration
mlt_properties_set_double( filter_props, "_previous_gain", gain );
mlt_properties_set_position( filter_props, "_last_position", current_position );
mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
// Ramp from the previous gain to the current
gain = previous_gain;
// Apply the gain
if ( normalise )
{
int16_t *p = *buffer;
// Determine numeric limits
int bytes_per_samp = (samp_width - 1) / 8 + 1;
int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
for ( i = 0; i < *samples; i++, gain += gain_step ) {
for ( j = 0; j < *channels; j++ ) {
sample = *p * gain;
*p = ROUND( sample );
if ( gain > 1.0 && normalise ) {
/* use limiter function instead of clipping */
*p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
}
p++;
}
}
}
else
{
float *p = *buffer;
for ( i = 0; i < *samples; i++, gain += gain_step ) {
for ( j = 0; j < *channels; j++, p++ ) {
p[0] *= gain;
}
}
}
return 0;
}