Since its invention by C. Rice and E. Kellogg in 1925 [1], the electrodynamic loudspeaker has been more than widely exploited in the audio industry. But if we come to discuss the means of predicting its acoustic behavior, it is rather when L. Beranek published his book Acoustics in 1954 [2][3] that the first models (based on the principle of Lumped elements [4]) were popularized. Among the electroacoustic parameters listed by Beranek in his book (also formulated by A. Thiele and R. Small at the same era), here are the fundamental ones:

       – 𝑅𝑒, the electrical resistive part of the voice coil, in ohms,

       – 𝐿𝑒, the electrical inductive part of the voice coil, in millihenries (often at 1k Hz),

       – B.l, the product of magnet field strength in the voice coil gap and the length of wire in the magnetic field, in tesla-meters,

       – 𝑀𝑀𝑆, the mass of the assembly “diaphragm / voice coil”, the mass of half of the outer suspension and the mass of air particles around the diaphragm (that becomes non-negligeable when dealing with small sized speakers), in kilograms,

       – 𝑅𝑀𝑆, the mechanical resistance of suspensions, in Newton-second per meters,

       – 𝐶𝑀𝑆, the total mechanical compliance of suspensions, in meter per Newton,

       – 𝑆𝐷 (mentioned in the form of the radius “a” in the Beranek’s book), the projected area of the speaker diaphragm.

In his book, Beranek also helped popularize the electrical network that defines the electro-mechano-acoustic behavior of such a loudspeaker. The diagrams in Figure 1 present three loops, dealing respectively with the electrical, mechanical, and acoustic domain of the transducer.

The field of haptics delves into the sense of touch through our mechanoreceptors under the skin. It’s used across various applications to convey information or sensations to the user. This ranges from simple vibrational feedback of a control button to complete immersion in a realistic tactile environment, enabling the perception of textures, shapes, and resistances, thereby enriching the sensory experience.

Originally inspired by telegraph and telecom domains, the audio industry has seen first uses of the audio equalization with persons like W. Mashall Leach, JR into his article Active Equalization of Closed-Box Loudspeaker Systems [1]. From the work exposed into his paper, a particular assembly of 2nd order active high-pass filters (HPF) is used to extend the low frequency response of a speaker system.

Nowadays findable in libraries of any real time signal processing solution, this simple type of filter is very easy to implement and tune since it can be controlled by only two parameters: the central frequency f and the quality factor Q.

Considering the below example illustrated in Figure 1, we can observe how much one HPF of f = 110 Hz and Q =2.2 extends the response of the presented enclosure.

There are various components that are included in Hearing Aid devices . The key components that are inside every hearing aid device are a microphone, DSP (Digital Signal Processor), speaker (commonly called a receiver within the industry), a battery, and a shell that encloses all the components.

Onsemi serves the Hearing Aid market with DSPs, and electronic peripherals such as EEPROMs, radio ICs, and power management components useful for integrating rechargeable batteries.

Onsemi’ solutions are offered as entirely programmable by the customer, called open-programmable products, and fully featured turnkey solutions with algorithms to load on the device, called Preconfigured Suite products.

Let’s dig out more about onsemi Ezairo series DSPs and their so called Preconfigured Suite (Pre Suite).*

Hardware overview :

Introduction and Working Principle

Originating from a setup created by Hermann von Helmholtz in the 1850s for determining sound pitches, this resonator consists of two main parts: a cavity and a neck, structured together as shown in Figure 1. When excited by an acoustical wave, the system will resonate at a frequency 𝑓₀, as given by the formula below, thereby creating an aero acoustic source.

When integrating a speaker into a device, it is essential to isolate its front radiation space from that of the rear. This is done in practice by building an enclosure dedicated to the transducer.

However, by doing so the rear volume of air applied to the speaker becomes an acoustic load acting as a high-pass filter on the system response: the smaller the volume, the more the low frequencies are filtered.

Two well-known techniques in the industry for recovering some low frequencies are the Bass-Reflex (BR) and the Passive Radiator (PR), respectively invented by A. L. Thuras in 1932 and H. Olson in 1935. These two techniques are called “passive” in the sense that they do not require additional electrical energy.

As illustrated in Figure 1, a BR system is simply made up of a tube (also called port), often cylindrical and integrated into the enclosure pierced for this purpose. In this configuration, the “cavity + port” assembly together forms an acoustic resonator identical in all respects to a Helmholtz one. If well-tuned, this resonator, driven by the loudspeaker, will radiate sound, thus extending the low frequency response (as in the example reported in Figure 3).

A PR system, shown in Figure 2, is based on the same resonator principle, but involving a diaphragm instead of a port. Because this diaphragm has suspensions, a notch (resulting from the “suspensions-diaphragm” assembly resonance) appears in the extended frequency region, thus causing a steeper roll-off than for a BR solution.

 When it comes to discussing pros and cons, this notch property of PR systems typically degrades transient response. On the other hand, BR solutions often bring non-linearities, resulting from air turbulence occurring in the port.

Among the many advantages and disadvantages that can be found in the literature, we will notice that BR solutions remain cost-effective and very easy to design, while PR systems become relevant when the size of the enclosure decreases, in addition to offer total sealing against dust and water.

In noisy environments, accurately picking up sound from a specific location is often a challenge. Since using conventional directive microphones may sometimes not be adequate or sufficient, beamforming techniques are nowadays widely considered in the audio industry. Focusing on 1-dimensional arrays, two well-known arrangements, named Broadside and End-Fire, simply consist of multiple omnidirectional microphones separated by a certain distance (fixed in the simplest cases). For the first configuration (illustrated in Figure 1), microphones are configured perpendicularly to the incoming wave direction, and the simplest audio operation can just be a summation (fixed array case). In the second configuration illustrated in Figure 2, microphones are installed in the same direction to the incoming wave. Audio operations are in this case a delay, a phase reverser, and a sum.

As we work more and more with multi-microphone networks in our laboratories, such as beamformers, we often want to control acoustic behavior of all the microphones at the same time. One method to accomplish this is a loudspeaker system relatively far from the microphone network so that all wavefronts in the bandwidth of interest are seen as planar by the network. (Figure 2 against Figure 1). In this configuration, all the microphones would receive incoming waves with the same acoustic energy and identical phase. However, depending on the frequencies of interest, this condition may require installing the loudspeaker system so far away from the microphone network that energy for correct measurement is not possible. An alternative way of building plane waves is by using a closed cavity (Figure 3), where one of the ends is connected to the loudspeaker system and the other connected and sealed to the microphone network in test.

When listening to music with a speaker system, we want perfect sound reproduction regardless of where we are in a room. However, depending on the type of speaker and the geometry of the speaker enclosure, there can be a large range of considerations to optimize the design.
This is one of the reasons why it is fundamental to properly quantify the behavior of a speaker system in an appropriate acoustic environment.
This analysis can be achieved by performing sound pressure level measurements at different angles around the system.
The step angle between each measurement will determine the accuracy : the smaller the angle step, the greater the precision.
Since this measurement is sensitive to disturbances coming from directions other than the speaker under test, it must be performed in an anechoic environment.
The results are usually reported in a polar pattern, showing the attenuation of the sound perceived by a person located at different points around the speaker enclosure. At 0°, the listener is located directly in front of the speaker and therefore perceives 100% of the sound (0 dB attenuation). As the listener’s location around the audio source changes, they perceive less sound energy, depending on the acoustic frequency.