Parametric sound is a fundamentally new class of audio, which relies on a non-linear mixing of an audio signal with an ultrasonic carrier. One of the key enablers for this technology is a high-amplitude, efficient ultrasonic source, which is referred to here as an emitter or transducer. Ultrasonic emitters can be created through a variety of different fundamental mechanisms, such as piezoelectric, electrostatic, and thermoacoustic, to name a few. Electrostatic emitters are generally capacitive devices consisting of two conductive faces with an air gap, where at least one of the conductive faces has a texture that is critical to the functionality of the emitter.
Non-linear transduction results from the introduction of sufficiently intense, audio-modulated ultrasonic signals into an air column. Self-demodulation, or down-conversion, occurs along the air column resulting in the production of an audible acoustic signal. This process occurs because of the known physical principle that when two sound waves with different frequencies are radiated simultaneously in the same medium, a modulated waveform including the sum and difference of the two frequencies is produced by the non-linear (parametric) interaction of the two sound waves. When the two original sound waves are ultrasonic waves and the difference between them is selected to be an audio frequency, an audible sound can be generated by the parametric interaction.
Parametric audio reproduction systems produce sound through the heterodyning of two acoustic signals in a non-linear process that occurs in a medium such as air. The acoustic signals are typically in the ultrasound frequency range. The non-linearity of the medium results in acoustic signals produced by the medium that are the sum and difference of the acoustic signals. Thus, two ultrasound signals that are separated in frequency can result in a difference tone that is within the 60 Hz to 20,000 Hz range of human hearing.
Like conventional audio systems, ultrasonic audio systems can suffer from distortion caused by the phenomenon known as clipping. Clipping is a form of waveform distortion that cuts off the peaks and troughs of the audio waveform when the signal is driven beyond the capacity of the amplifier. Clipping occurs when an audio signal exceeds a maximum value allowed by the audio system. The signal beyond the capability of the system is cut off, or clipped, resulting in distortion of the audio signal and the creation of nonlinear distortion products, such as unwanted harmonics related to the input.
Digital signal processing (DSP) has been applied extensively to audio, including to perform functions such as compression, equalization, surround sound, pitch control, etc. Many of these algorithms can apply gain to an input signal and the resulting output can exceed the maximum signal level allowed by digital to analog conversion after processing. The resulting signal is ‘clipped’ whereby any signal attempting to exceed the maximum value is reduced, or clipped at the maximum value. Similarly, there is a minimum value whereby signals below this value are raised. To illustrate this, consider a system having its output limited to a range between −1 to 1 volts. Any value below −1 is raised to −1, and any value above 1 is dropped to 1. An example of this clipping is shown in FIG. 1. As seen in this example, an 1 kHZ input tone with an amplitude of 2 volts peak-to-peak (top of Figure) is clipped at +/−1 volt (bottom of Figure).
The resulting spectrum is shown in FIG. 2. Before clipping the spectrum is a pure tone with one frequency component (top of Figure). After clipping, odd (3rd, 5th, 7th, etc.) appear and significantly color the audio experienced (bottom of Figure).