Audio

syne.studio uses a custom audio engine built from the ground up in Rust. This enables us to perform advanced, high quality digital signal processing (DSP) in your web browser at near-native speeds.

Sample rate is the number of audio samples per second contained in an audio signal. A higher sample rate will increase the highest frequency that the audio can contain (the nyquist frequency), a lower sample rate can greatly improve CPU performance for DSP to prevent audio dropouts, and reduce storage size and memory usage.

A 44.1 KHz sample rate is considered high quality, and is used on CDs. Some may prefer 48 KHz, since it raises the nyquist frequency from 22.05 KHz, which is lower than some people's top audible frequency, to 24 KHz, which is higher than any listener should be able to hear. Sample rates higher than 48 KHz are typically only desirable for specific use cases (if you don't know why you need them, you probably don't need them).

The sample rate of Syne's audio engine is determined by the default sample rate of your browser. In some cases (e.g. pipewire on Linux), this sample rate may be adjustable.

Firefox does not support customizing the playback sample rate of the project without introducing issues with audio recording, but in the future Syne will support customizing the sample rate within the application on Chrome and other browsers that support it.

Racks are the primary unit of organization for a project.

Each rack can have at most one instrument, which turns MIDI notes from the rack's clips into audio output. They can also have any number of audio effects that take the output signal from the rack's instrument and/or audio clips and transform it one after another. Finally, the audio signal is sent to the rack's mixer (controlled via the Rack Controls), and on to the Main Bus.

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Audio Effects in syne.studio are stereo-in, stereo-out plugins that a rack's audio passes through before reaching its output. Effects can be edited from the Rack Editor. All audio effects settings are project-level controls.

Syne audio effects are all built in-house, and optimized for the best possible performance and DSP quality.

Bit Crusher can reduce the bit depth of the audio signal and/or reduce its sample rate, introducing randomness and imparting a digital lo-fi quality to the audio signal.

The top row contains knobs to control the downsample rate, as well as the bit depth.

The bottom row contains a knob to add variability to the downsampler section of Bit Crusher. Increasing variability will affect the characteristic sound of the downsampler, changing the defined harmonics into a noisier profile. Finally, a dry/wet knob balances the original signal with the bitcrushed one.

The compressor is used to reduce the dynamic range of the audio signal. It's a broadly useful effect, and especially good for mixing or sculpting transients. Learn more about the basics of compression here!

The left slider controls the threshold at which the compressor is activated, in dB.

The top row contains the attack and release knobs, which control the speed at which the compressor reacts to the incoming audio signal.

The middle row contains the ratio and knee knobs. Ratio denotates the compression factor after the audio signal's volume surpasses the threshold. Knee, represented in dB, will impart a soft knee on that compression factor, smoothing the gain reduction as an audio signal approaches the threshold. This makes the resulting compression sound more natural and less aggressive.

Finally, the bottom row contains the input gain and output gain knobs, with values represented in dB. These will add or remove gain to the input or output respectively.

Delay is used to repeat the audio signal after a certain delay, potentially multiple times.

The top row contains a knob to control the length of the delay, and a button to switch between units of seconds and measures. When using measures, the delay length will stay synchronized with any changes to the project's BPM.

The bottom row has a feedback knob, which controls how much of the delay's output is fed back into it's input. This creates a repeated delay that gradually fades out with each repetition. Finally, there is a dry/wet knob which balances the original signal with the delayed one.

Distortion is used to add harmonic content to an audio signal via saturation.

Multiple models are available from the drop-down box, each with their own harmonic characteristics.

The drive knob increases the gain coming into the effect in dB, which will increase the overall amount of distortion. Additionally, the makeup knob adjusts the gain coming out of the effect in dB, to account for drive and provide gain staging.

Finally, the dry/wet knob balances the original signal with the distorted one.

An EQ (parametric equalizer) combines multiple filters in order to make precise boosts or cuts to an audio signal.

The frequency content of the audio output from the equalizer is represented in the main spectrograph view, with the x-axis representing frequency, and the y-axis representing amplitude in dB. As filters are added, a line representing the frequency response curve will show the sum-total effect of the filters.

A new EQ instance contains no filters by default; double click within the Spectrograph to add new filters. Within the spectrograph, the filters are represented by draggable circle icons, with parameters mapped to the x and y axes. Clicking on one of these icons displays the filter’s controls to the right of the spectrograph (see Filter for more info).

The spectrograph can be resized along its x-axis by clicking and dragging from the right edge of the spectrograph.

Filters are used to selectively boost or cut specific frequency ranges of an audio signal.

The top row contains a knob to control the cutoff or center frequency, and a knob to control the q, or quality factor. A higher quality factor results in a narrow, sharp curve, while a lower Quality Factor results in a wider, more shallow curve. An additional morph knob is available for the morph filter type, which dynamically morphs the filter between Lowpass, Peak, and Highpass.

The bottom row contains a filter type selector to change the type of filter. For most filter types, an additional stages knob will add additional stages to the filter, resulting in a steeper slope per octave.

Polyshev is used to add controlled harmonic content to an audio signal. Learn more about chebyshev polynomial waveshapers here!

dry is the original, unprocessed audio signal. odd polynomials (3rd, 5th, 7th...) have a hollow character, while even polynomials (2nd, 4th, 6th...) add warmth and richness, similar to tube saturation. For each control, -100% represents a phase-flipped signal at full volume, 0% is silent, and 100% is the original phase at full volume. The relationship between the fundamental and its odd and even polynomials alongside phase-flip has a drastic affect on the resulting sound.

The color controls how many polynomials are included: lower values limit the effect to just the first few harmonics, while higher values allow more overtones through, increasing the complexity of the resulting harmonic content.

attenuation controls how the volume of the signal coming into the waveshaper in dB. Polyshev uses a type of non-linear distortion, with many different sounds achievable through controlling the amount of saturation occuring within the waveshaper.

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