Vocal synthesis using sine: Difference between revisions

=== '''Restoring Low Frequency Clarity''' ===

[[File:Amenbreak drum sample before conversion using Liam's FFT.wav|thumb|An amenbreak drum sample audio file before conversion using Liam's FFT]]This algorithm is highly effective at replicating the mid-to-high frequencies of an sound file, but depending on the chunking frequency in the Advanced Options panel, the lower frequencies will tend to be grainy and noisy, and will also lack clarity. A possible way to mitigate this limitation is to time-stretch your sound file using an audio editor, such as Audacity. Using the Effects tab found on the topmost panel, navigate to Pitch and Tempo (for Audacity versions 3.3+), then click on the Change Tempo option, while having the audio clip selected. Change the Beats per Minute field to "from 16 to 1" (either that or another power of 2 value such as 8-1 or 32-1, higher values give more frequency clarity, but will significantly inflate the note count). It's also essential to make sure that the high quality stretching checkbox is marked, otherwise Audacity will use the default algorithm of tempo-stretching, which is very low quality. The purpose of stretching the sound file is to allow more data for chunking in the FFT converter, which will increase frequency accuracy and also retain time accuracy. Return to the converter, open the time-stretched sound file, and divide both the minimum and maximum chunking frequency by the value which the audio has been time-stretched (e.g. 16). Run Liam's FFT converter, this time with the time-stretched audio file. Once it is finished, import the output *.sequence file into Online Sequencer and use the console to stretch the notes by a factor of 1/N, with N being the value used to time stretch the sound file. The resulting sequence should have clearer low frequencies, and less graininess or noise.[[File:Amenbreak drum sample after conversion and exported to WAV.wav|thumb|An amenbreak drum sample audio file after conversion using Liam's FFT, rendered to *.wav, using the specified method to enhance similarity to the original file]]<nowiki>*</nowiki>Additional Note: To make the output sequence sound more like the original audio file, try using a windowing function (e.g. Blackman), a high extra-detune value (e.g. 4800), a low minimum note volume (e.g. 0.0025), stereo (if needed), increasing the number of microtones (e.g. 4 microtones), and perhaps increasing the overall output volume (e.g. 4). The example settings achieve a result that sounds rather close to the original sound file, although it will be very note dense, so it's best to export the sequence as an audio file to listen to it without lag or cutouts.