Difference between revisions of "Cochlear Implants"

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==Pre-emphasis filter==
 
==Pre-emphasis filter==
 
[[file:Pre_emphasis_filter.png|thumb|an example of a pre-emphasis filter]]
 
[[file:Pre_emphasis_filter.png|thumb|an example of a pre-emphasis filter]]
The pre-emphasis filter is meant to enhance speech recognition by lowering the weight of the lower frequencies in sound. As a consequence the loudness measured in dB of the sound after filtering will be lower than before the filtering. The loudness difference will depend on the exact shape of the filter as well as the spectral content of the sound. When presenting pink noise to the filter, as applied in AB devices, the overall loudness of the signal is diminished by about 10 dB, but, when presenting speech, the loudness will be less effected due to their different spectral content.
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The pre-emphasis filter is designed to enhance speech recognition by attenuating lower frequencies in sound, which are less critical for understanding speech. This attenuation helps emphasize higher frequencies, where important speech information, like consonant sounds, is more prominent. As a result, the overall loudness of the sound, measured in decibels (dB), is typically reduced after filtering. The degree of loudness reduction depends on both the filter's characteristics and the spectral content of the input sound.
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For example, when pink noise is processed through the pre-emphasis filter, as applied in Advanced Bionics (AB) devices, the loudness is reduced by approximately 10 dB. However, when speech is processed, the reduction in loudness is less pronounced due to the different spectral content of speech, which has more energy in the higher frequency bands where the filter has less impact.
  
 
==Automatic Gain Control==
 
==Automatic Gain Control==

Revision as of 13:39, 15 August 2024

Introduction

A hearing aid with a cochlear implants translates incoming sound into electrical signals that are directly stimulating the nerves in the cochlea. The process of converting the sound waves into electrical current is done by a sound processor outside of the ear. The steps are the following:

CI signal processing scheme
  • Microphone picks up sound
  • ADC converts analog signal to digital signal
  • Pre-filtering is applied for speech emphasis
  • Automatic Gain Control (AGC) is applied
  • Signal is split into frequency bands by a filter bank
  • Hilbert Envelopes (amplitudes) are calculated per band
  • Signal amplitudes per band are mapped to current amplitudes
  • Current amplitudes per band are convoluted with spiking patterns
  • Current steering distributes the current of a single band over multiple electrodes
  • The resulting signals sent to the electrodes

Pre-emphasis filter

an example of a pre-emphasis filter

The pre-emphasis filter is designed to enhance speech recognition by attenuating lower frequencies in sound, which are less critical for understanding speech. This attenuation helps emphasize higher frequencies, where important speech information, like consonant sounds, is more prominent. As a result, the overall loudness of the sound, measured in decibels (dB), is typically reduced after filtering. The degree of loudness reduction depends on both the filter's characteristics and the spectral content of the input sound.

For example, when pink noise is processed through the pre-emphasis filter, as applied in Advanced Bionics (AB) devices, the loudness is reduced by approximately 10 dB. However, when speech is processed, the reduction in loudness is less pronounced due to the different spectral content of speech, which has more energy in the higher frequency bands where the filter has less impact.

Automatic Gain Control

Automatic Gain Control (AGC) in cochlear implants serves two primary purposes:

  • It maintains speech levels near the most comfortable listening level for the CI user.
  • It rapidly reduces the gain when very loud sounds are detected to prevent discomfort.

The AGC achieves these goals through two systems:

  • Averagers (fast and slow): These use buffers to average sound levels over specific time windows. Fast averagers respond quickly to sudden changes in sound level, while slow averagers handle more gradual changes. When the average level exceeds a predetermined threshold, the averagers calculate the excess loudness, triggering the compression system.
  • Compression: When excess loudness is detected by the averagers, the compression system reduces the gain according to a pre-defined compression function. This function usually applies more compression to louder sounds, effectively narrowing the dynamic range in the output signal and ensuring that softer sounds remain audible while protecting the user from loud sounds.

Fast and Slow averagers

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Compression

See Dynamic_range_compression on wikipedia for more background info. %todo

Band filtering

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Hilbert Envelopes

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Spike patterns

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The mapping function

The mapping function (according to Advanced Bionics)

The mapping function translates the Hilbert Envelope Amplitudes for each channel to electrical signal amplitudes.

Different manufactorers use different mapping functions:

Type of Mapping Formula Manufacturer
Logarithmic Mapping I = G * log(L) + C Advanced Bionics
Power Law Mapping I = G * L^α + C Cochlear
Linear Mapping I = G * L + C MED-EL

G = proportionality factor or slope (sometimes confusingly called gain)
C = some constant

Sometimes different constant are used below and above the kneepoint.

Advanced Bionics Mapping Function

Following Advanced Bionics we can fill in some details of their mapping function:

I = ((M-T)/IDR) * (L - KNEE + IDR + GAIN) + T

where

  • I is the electrical output in current units
  • M is the most comfortable level in current units (CU)
  • T is the threshold level in current units (CU)
  • L is the loudness signal (Hilbert Envelope amplitude) in dB SPL
  • KNEE is the kneepoint of the compression function in dB SPL
  • IDR is the input dynamic range in dB SPL
  • GAIN is the channel gain in dB SPL

N.B. Current units (CU) is defined differently by different manufacturers.

It follows from the equation that when L = KNEE - GAIN result in I = M, so the electrical output I is right at the M-Level. Since the GAIN is set to zero in most cases, the output is at the M-level when L = KNEE.

Terminology of Critical Points in the Mapping Function

In scientific literature, three key levels are recognized: 1. hearing threshold, 2. comfortable level, and 3. pain threshold, each denoted in various ways.

The following abbreviations are commonly used to describe specific levels of electrical stimulation:

  • T, THR, or THL: These stand for Threshold Level, or Threshold Hearing Level. This is the lowest level of electrical stimulation that the user can perceive.
  • M, MCL, or C: These abbreviations stand for Most Comfortable Level, Maximum Comfortable Level, or Comfortable Level. This level refers to the point at which sounds are comfortably perceived—neither too loud nor too soft.
  • USL, MSL, UCL, or LDL: These stand for Upper Stimulus Level, Maximum Stimulation Level, Uncomfortable Loudness Level, or Loudness Discomfort Level. This level indicates the point at which sound becomes painful or uncomfortable for the user.

The terminology can vary depending on the manufacturer. For example, for the comfort level in clinical software:

  • Advanced Bionics uses 'M-level',
  • Cochlear uses 'C-level',
  • Med-El uses 'MCL'.

There can also be some variation in the precise definitions of these terms.

The term "Maximum Comfortable Level" is less commonly used and can sometimes refer to the highest level that is still comfortable, just below the pain threshold. This usage can be confusing and should be avoided where possible.

Current Steering

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