float *output_buffer = mlt_properties_get_data( properties, "resample.output_buffer", NULL );
int i;
+ if ( output_rate == 0 )
+ output_rate = *frequency;
+
// Restore the original get_audio
frame->get_audio = mlt_properties_get_data( properties, "resample.get_audio", NULL );
// Get the producer's audio
mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+ //fprintf( stderr, "resample_get_audio: output_rate %d\n", output_rate, *frequency );
+ // Return now if now work to do
+ if ( output_rate == *frequency )
+ return 0;
+
// Convert to floating point
for ( i = 0; i < *samples * *channels; ++i )
input_buffer[ i ] = ( float )( (*buffer)[ i ] ) / 32768;
(*buffer)[ i ] = lrint( 32768.0 * sample );
}
}
- //else
- //fprintf( stderr, "resample_get_audio: %s\n", src_strerror( i ) );
+ else
+ fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
return 0;
}
}
else
{
- //fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
+ fprintf( stderr, "filter_resample_init: %s\n", src_strerror( error ) );
}
}
return this;