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 English - International

Automatic Gain Control Provides
a Carefree Listening Experience

The Challenge

One of the biggest challenges facing any cochlear implant system is that of compressing a widely varying loudness spectrum into a very small electrical dynamic range. The ear with normal hearing can generally process loudness inputs from 0 to 120 dB HL with little difficulty. Of course, as we approach our loudness "limit," we begin to experience negative effects, like ear discomfort or even pain. These cues tell us that the sound we are hearing is loud enough to cause permanent damage to our hearing, and they prompt us to somehow change the environment to protect our ears.

With electrical hearing, the picture is very different. Here, the ear with an implant generally has a dynamic range of 10-20 dB, and sometimes it is even as low as just a few decibels. This is a result of the processes that created the sensorineural hearing loss in the first place, and also of the way in which nerve fibres respond to electrical stimulation. All of this means that the cochlear implant system needs to be able to accurately represent subtle loudness cues, as rapidly as they change in the environment, across the wide range of normal hearing by compressing those cues into the very narrow dynamic range of electrical hearing. It is the subtle differences between energy peaks in a speech signal that let us decode that signal and give it meaning, so we need to represent those cues accurately, but also as rapidly as the subtle changes in running speech, or in a musical piece.
 
Note the many rapid loudness variations as the word “cochlea” is spoken.

Possible solutions

A system could, of course, use only a single-stage compression scheme – but the hearing aid literature tells us that speech understanding with this type of system is rather poor. It’s poor because the system is trying to do two things in one step: while it is trying to compensate for the overall volume of the input signal (which has to happen relatively slowly), it can’t react quickly enough to sudden changes in the sound level. The result is that the user can perceive “dead points” or moments when sound cuts out completely. Therefore, a single-stage compression scheme will not be sufficiently  able to manage overall sound level while still accurately representing subtle loudness cue changes that happen very rapidly.
The best solution is a two-stage compression design. The first stage, the dynamic step, is the one we will focus on here. This first stage is the Dual-Loop Automatic Gain Control (AGC). The second stage, which occurs later in the sound coding process, is a static step called maplaw. The maplaw is a compression parameter that is applied equally to all channels and determines where “soft,” “medium,” and “loud” sounds are mapped into the user's individually-measured dynamic range. The maplaw component of the coding strategy is customisable by the audiologist during the fitting session.
The dual-loop AGC – the first, dynamic, stage of the process – is a clever approach to accurately representing loudness variations and details in constantly-changing environments. Although the audiologist will often test a patient’s ability to hear in background noise, this test environment, in a sound-treated booth with a fixed presentation level for speech and a fixed level for noise, is not very representative of the real world. MED-EL’s dual-loop AGC, however, uses two monitoring systems to track and manage incoming sound based on the overall sound level – making the MED-EL system automatically adaptive to the environment.

How does dual-loop AGC work?

AGC systems all have one shared problem. They need to manage the overall sound level, but they also need to manage unexpected intense transient sounds in order to remain comfortable to the user. Those two things require completely different approaches. Overall sound level needs to be monitored slowly, but transient sounds come and go so quickly that a system with just one loop would miss them completely, resulting in uncomfortably loud stimulation for the user every time an unexpectedly loud sound occurs in the environment. If the AGC is built to react quickly, then the user perceives the sound “pumping in and out.” There is a fine balance between what type of attack time is needed vs. what can be perceived, and what type of release time is needed vs. what can be perceived, and the two can’t easily coexist peacefully in one circuit.

MED-EL resolves this paradox in the following way. All incoming sound passes through two peak detectors that co-exist side-by-side. One detector analyses rather slowly (on the order of several hundred milliseconds). The other analyses quickly. You can almost imagine the slow detector analysing at a relatively slow “rate,” checking in now and then to see what the overall level is. The fast detector is analysing very quickly – it is looking for loud, transient sounds.
Both of these detectors have the ability to change the gain of the microphone. Usually the slow detector is in control (even though both are always working). So if the overall sound level of the environment changes (as when walking from a quiet room to a noisy room), the slow detector adjusts the microphone gain to be sure the input is within the sound processing window of the implant system. But if the fast detector notices a sound that exceeds the slow detector by more than 6 dB in a very short time period, it quickly takes control, reduces the gain of the microphone (if necessary), and then gives control back to the slow detector. The slow detector then just picks up right where it left off - because the overall sound level didn’t really change. This allows the management of the transient sound to be handled without the user noticing any negative effects.

Taking a closer look

First, let’s look at the slow detector. The slow detector operates essentially as an automatic volume control. MED-EL implant users have the ability to control volume with an adjustment button on the processor or with the Fine Tuner (depending on the processor style). However, nearly all patients report that they do not feel the need to adjust volume as they go about their day, regardless of the changing environments that they experience. That’s because the slow detector is constantly focusing on the sound level in the present moment. There is no perception of anything adjusting in the background – just effortless movement between one environment and another without the patient being required to make adjustments manually to support the processor.
The slow detector is in control the majority of the time, until a loud transient sound is detected by the system. At this point, control shifts to the fast detector. The fast detector’s job is to identify the transient, compress it very rapidly so that the patient doesn’t experience discomfort, and then hand control immediately back to the slow detector.
The end result of this process is shown here (above). On the first line, you see the original sound signal as it changes over time. It is a speech signal, and you’ll notice a loud sound obscuring the speech signal in the first moments of the tracing. The middle line shows the result of a standard single-loop compression system. The transient sound is identified and rapidly compressed, but the system must take a relatively long time to release the compression to avoid having the patient hear a “motor-boating” or “pumping” sound. In fact, it appears that the speech signal doesn’t return to its normal loudness level until more than a full second has passed.

The third line, however, shows the result of a dual-loop AGC. The transient sound is managed nearly instantaneously, and the management of the overall sound level is returned back to the slow detector within just a fraction of a second.
 

The result

The end result for the patient is unique – the patient just puts on the system and goes about their day. The dual-loop AGC manages the varying environments that the patient encounters, without manual input, over a 75 dB input range (accurately processing inputs from 25 – 100 dB SPL). In the patient’s sound experience, soft sounds are appropriately soft and loud sounds are proportionally loud, but specific program or setting changes to accommodate the differences between noisy environments, music, quiet environments or speech as the primary signal are simply not necessary.

The dual-loop AGC design increases the robustness of the signal when high level transient sounds are present, which means a better chance at better speech understanding for the user, even in the most adverse listening conditions. To view a study that attempted to recreate the varying listening conditions found in the real world and assess the performance of the front-end preprocessing of all three cochlear implant systems in this setting, click here.
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