Our low latency audio source separation technology removes the interfering noises that make speech communication difficult. The same technology that improves speech for hearing assistance will remove interference from your VoIP audio stream, allowing your audio compression codecs to do a better job of compressing the audio without introducing the unpleasant artefacts (such as pumping and noise masking) that make the transmitted audio difficult to understand. Our technology offers the following key features necessary for any practical product:
The array geometry is not critical - we can work with 3 to 8 microphones with any spacing from 2cm to 30cm apart.
There is no calibration or algorithm training requirement, minimising your production costs.
The algorithm has a low latency (<10ms) mode for time critical applications.
The algorithm automatically compensates for microphone occlusion and gain mismatch.
The algorithm gives consistent, natural sounding performance across a wide variety of real world conditions.
The highly efficient algorithm and coding is designed to obtain the best possible performance from the available processing power.