Synthesize Using Samplers


Sample-based synthesis is a form of audio synthesis that mainly involves use of samplers, either hardware or software. The main difference between this form of audio synthesis and the others is that it uses sampled sounds or instruments instead of having oscillators with fundamental waveforms such as saw, square, triangle, etc.


Back In The Day


In the beginning of the second half of the 20th century, before digital sampling was used practically , machines like Mellotron were using analog tape decks in order to playback sampled sounds. Later on, with the birth of more powerful samplers in the 1970`s and 1980`s, sample-based synthesis idea evolved. The concept behind this form of synthesis was to emulate real instruments. This is done by recalling actual samples of these instruments upon striking the keys on the keyboard.

These samplers were rather expensive back then and they could only offer only scarce sample rate and bit depth, often resulting in grainy and aliased sound. Not to mention that they were limited by the expense of memory and therefore utilized the shortest possible length of the sampled sounds. Moving forward to recent times, computers became capable of providing much larger memory and processing speed, so we were introduced to the software samplers with a lot more editing possibilities.


Main Aspects


So, there are 3 main issues to address in sample-based synthesis:


1. Looping


Looping extends the duration of sampled sounds played by musical keyboard. If the musician holds down a key, the sampler should scan “ seamlessly” through the note until the musician releases the key. Furthermore, this is done by specifying beginning and ending  “loop points“ in the sampled sound. After the attack of the note is finished, the sampler reads repeatedly through the looper part of the wavetable until the key is released.

Most of the latest samplers provide automatic methods for finding prospective loop points. One of these methods is to perform pitch detection on the sampled sound. This means that the pitch detection algorithm searches for repeating pattern in the wavetable that indicate a fundamental pitch period. Therefore, the pitch period is the time interval that spans one cycle of a periodic waveform. Once the pitch has been estimated, the sampler suggest a pair of loop points that match same number of the pitch periods in the waveform. This kind of algorithm tends to create smooth loops that are constant in pitch.

2. Pitch shifting


Indeed, it may not be possible to store every note played by an acoustic instrument in an inexpensive sampler. This is because these samplers store only every 3rd or 4th semitone and obtain intermediate notes by shifting the pitch of a nearby stored note. If you record a sound into a sampler memory  and play it back by playing different keys, sampler carries out the same pitch shifting technique. The side effect of the simple pitch shifting is that the sound duration increases or decreases, depending of the key pressed. There are two methods of the simple pitch shifting to be mentioned.

Method 1:


Varying the clock frequency of the output DAC changes the sampling rate. That changes the pitch up or down and changes the duration.

Method 2:


Sample rate conversion ( resampling the signal in the digital domain ) shifts the pitch inside of sampler and allows the playback at a constant sampling rate for all pitches.


3. Data reduction


A way to reduce data stored in a sampler is to to limit samples resolution or quantization. Another way would be done by reducing sampling rate. This diminishes a number of samples stored per unit of time, at the cost of shrinking the audio bandwidth. In addition, a more sophisticated way of data reduction starts from an analysis stage.  It stores sound in a data reduced form, along with control functions that approximately reconstitute it.


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