Codec

Intro

Codec is a bit crusher / digital sound degrader that works by using a real-time audio compression codec made for doing voice communications over the internet. It works with the VST3, AU, and AAX plugin standards on modern versions of Windows and macOS. This manual serves as a full explanation into every parameter and system featured within the scope of the plugin. It will be updated along with the program itself.

Be sure to check out the Common Features page to learn about capabilities that every Lese plugin has.

Controls

Loss

The "Loss" control changes the rate at which "packets" will be dropped in real-time. The control is measured as a percentage, i.e. the percent chance that a packet will be dropped. When a packet is in it's "dropped" state, the compression codec will attempt to "conceal" the packet loss until so many packets are being lost that it can no longer attempt to conceal it properly.

Loss Mode

The three buttons to the right of the Loss knob adjust the style of packet loss. Random is a totally random chance of packet drop, Smooth is a smoothed out randomness value (so the likelihood of loss can slowly increase or decrease over time), and Repeat "holds" packets by using the same packet over and over if a packet is considered to be lost.

Bitrate

Changes the bitrate of the encoding, and thus the size of the packet (internally). This value is measured in kilobytes. Decreasing it below 32k starts to introduce a lot of easily noticeable compression artifacts, and the bitrate can be brought all the way down to 2k.

Algorithm Switch

The switch to the right of the bitrate control swaps between two different algorithm modes, the primary one (and default) being for voice signals, and another being for music. These can be set to whichever you prefer of course, but changing between these can add a bit of subtle differences as both algorithms have different strengths and weaknesses. Swapping between the two algorithms will cut out the audio momentarily.

Bandwidth

Adjusts between the different bandwidth modes in the internal audio codec. When "auto" is selected, the cutoff frequency for the bandwidth adjusts dynamically based on the codec's bitrate, to attempt to filter out harsh artifacts, this can be adjusted so that the bandwidth is set to a constant value, including 4, 6, 8, 12, and 20 kilohertz.

Crunch

The various crunch controls add an option to add a bunch of additional amplitude to the audio prior to being encoded & decoded, and then having the same system perform inverse filtering on the audio after decoding is complete, so that distortion artifacts can be added to the audio, without additional volume. The Crunch control adjusts the magnitude of this filtering operation.

Width

Adjusts the bandwidth, or Q of the filtering operation.

Frequency

Changes the center frequency of the operation. Using the encoder & decoder can introduce a bit of latency, so it is advisable to adjust the frequency slowly over time. Unless you like weird stuff.

Mix

Adjusts the dry/wet mix of the audio signal. The dry signal is delay-compensated to match with the latency that is introduced in the internal encoding & decoding algorithms.

Visualizer

Codec's visualizer is just for a cool effect. It reacts to the input audio you send into it. There is also an easter egg.