Noise Suppression

Capsule noise, electric noise, electronic noise, and environmental acoustic noise are removed from microphone signals. Alternatively, also transducer/loudspeaker signals can be pre-processed to reduce noise in the playback. Audio artifacts are minimized by employing perceptual principles.

Noise attenuation up to 25 dB
Algorithmic delay 2 to 10 ms


For cost and other reasons, one is often limited by the use of only one microphone. This instantly converging technology attenuates reverb and other non-stationary noises from single-channel signals.

Convergence time instant
Reverb attenuation up to 20 dB
Algorithmic delay 2 to 10 ms


Our patented multi-microphone beamforming and non-stationary noise suppression algorithms enable a new level of conversation comfort. The use of this technology with two microphone capsules enables the removal of non-stationary environment noise, de-reverberation, and improved handsfree operation.

Direct sound attenuation up to 20 dB
Reverb attenuation up to 20 dB
Array type small, 2 or 3 microphones
Microphone type omni, pressure gradient
Algorithmic delay 2 to 10 ms

Acoustic Echo Control

Our patented advanced acoustic echo control solution scales for use in professional tele-conferencing systems, personal computers, tablets, and telephones. Its low computational complexity makes it suitable for use in handsets and mobile devices for standard and wideband handsfree voice calls. It scales for use in next generation high-end teleconferencing, supporting sampling rates of up to 48kHz and stereo/multi-channel audio.

Echo attenuation up to 80 dB
Audio I/O delay up to 500 ms
Delay estimation continuous
Non-linearity modeling On/Off/Auto
Doubletalk yes
Initial convergence instant
Algorithmic delay 2 to 10 ms
Time-Jitter Control Up to 20% per frame
Saturation Detection On/Off

Gain Control

Important for handsfree, auto gain control (AGC) adjusts the microphone level to reach a desired target level. If available, our AGC interacts seamlessly with the hardware’s/ADC’s programmable gain, through a user callback function. A limiter makes sure that hard clipping never occurs.

A compressor/limiter, specifically developed for small loudspeakers and speech, reduces non-linearity and improves performance at high levels.

Fast gain control -12 to 12 dB
Slow gain control -20 to 20 dB
Hardware gain control full hardware range
Voice activity detection near-end and optionally far-end
Compressor On/Off, Ratio 1 to 20
Limiter On/Off
Algorithmic delay 0 to 10 ms

Sampling Rate Conversion

Often one has the problem that the ADC or DAC operate at different sampling rates than signals sent or received. Illusonic’s suite of low-complexity down-samplers, up-samplers, and re-samplers often come in handy.

Time Drift Control

Usually, the clocks of the incoming signals differ from clock of the local DAC. Or, the clock of the outgoing signal may differ from the ADC. Flexible time-drift control makes sure Illusonic Voice technologies work well, in realistic conditions.


A perfect loudspeaker or microphone does not have a flat frequency response, depending on device form factor, how it is built into a device, and other practical constraints. Illusonic can provide sophisticated software tools for optimising its fully parametric equalisers to fine-tune the effective frequency response of  loudspeakers and microphones in a device. Multi-point measurement enables robust optimisation.