Amp I Flashcards
Careful visualization of this area of the external auditory canal during otoscopy allows audiologists to identify the location of the second bend.
Anterior canal wall
Ideally, the canal length of an earmold impression will reach this depth to supply all the necessary information on anatomy an earmold manufacturer requires
2mm past the second bend
Combines 1:1 polyvinyl-siloxane parts of paste
Addition-cured silicone
Dimethyl-siloxane paste mixed with hardener
Condensation-cured silicone
Pre-measured acrylic powder and liquid formulas are rapidly mixed together
Methyl-Methacrylate
What is the primary reason for using high-viscosity impression material when making earmold impressions?
It supplies maximum stretch of the aperture
Which three case history questions must be asked immediately before every earmold impression is taken (even when you’re working with a well-established patient!)
Are you immunocompromised; have you ever had surgery on your ears; do you take blood thinners
To effectively assess the external auditory canal using diagnostic otoscopy, the audiologist may sit or stand. The viewing position and angle do not make a difference
False
Earmold impressions should not be taken when a perforation of PE tube is present
False
This type of otoblock allows the audiologist to create a longer impression of the external auditory canal with less patient discomfort.
Cotton otoblock that’s equal in size to the EAC entrance
You feel resistance during otoblock insertion suggesting you’ve chosen the correct size. If the resistance suddenly decreases as you move it more deeply in the canal it indicates
the patients ear canal widens beyond the 2nd bend
Open jaw impressions
Increase the size of the aperture
What would you do for PT with BTE in humid environments that sweats a lot?
High ingress protection rating
What does changing a vent size effect
Low frequencies and high frequencies
3 temporal resolution cues
What would not change with length of impression?
Standing waves in the ear canal
Explain how Electric mic transduces an acoustic signal into an analogous electric signal
The electret mic has an electret film applied to the backplate that is pre-charged with electrons to provide the voltage that is needed to transduce a signal. The acoustic signals arrive at the mic where they are transduced into an analog electric signal. The acoustic signal pushes against the diaphragm, decreasing the space between it and the pre-charged electric backplate. The pushing in of the diaphragm to the back plate condenses the particles causing a positive analog signal. When the diaphragm pulls back out it causes rarefaction of the particles and a negative analog signal. This push and pull of the diaphragm in sync with the acoustic sine wave against the pre-charged backplate creates the + and - analog electric signal in an electret mic
What are the limitations of electric microphones? How does MEMS overcome these problems?
The electret microphone requires a diaphragm and electron stability to maintain microphone sensitivity. Environmental factors, such as humidity, and temperature extremes can degrade the microphone. The diaphragm can absorb moisture which limits movement, and breakdown the adhesive required to hold it in place. Temperature extremes can cause electron loss from the electret backplate
The MEMS mic overcomes these concerns by using a silicone disc for the diaphragm, so it does not rely on adhesives, and by the addition of a charge pump to replace lost electrons and maintain the electrical held.
A PT has digital HA. HA distorts when they have louder acoustic signals, like a symphony. Could be front end distortion. What is it and how is it caused
Front-end distortion describes a microphone limitation associated with its dynamic range. The microphone dynamic range is the difference between the mics noise Roor and the loudest signal it can collect. 16-bit digital hearing aids can ahly collect signals up to 96 dB before going into saturation. Signals louder than this will be clipped adding distortion to the output signal.
Real world snr is ______ because of head shadow and polar plots
2-3
If Signal was 76 and noise 74 what is SNR?
2 dB
Greatest directivity index -
hypercartiod
There is a blockage at the back port. Why can this mic no longer make nulls and no longer function as a directional mic but an omnidirectional mic
When something plug the port, the sound coming in from the back doesn’t reach the diaphragm which doesn’t cause an out of phase signal which means no directionality occurs
How does a telecoil create an analog electric signal
A telecoil transduces an electromagnetic signal instead of an acoustic signal by applying the induction principle. A small magnet sits inside a tightly wound copper coil creating an electromagnetic field. The electromagnetic signal pushes and pulls the magnet. This movement inside the copper coil produces a positive and negative electrical current flow analogous to the incoming electromagnetic signal.
directional microphone function which is true:
Directivity index reduces as diaphragm adhesive degrades
All of these answers are true
Directional microphone roll off increases as the external time delay decreases
Directivity index reduces when directional microphone ports are not parallel to the floor
all
Issues understanding at a bar. Lombard reflex increases the intensity of your friend’s voice in this environment. They speak at 76 dB SPL. Noise level is 74 dB. SPL What is the SNR
2
As the distance increases between an acoustic signal and microphone:
The arriving signal to noise ratio decreases
The arriving signal becomes softer but the signal to noise ratio does not change
The arriving signal to noise ratio increases
The arriving signal to noise ratio decreases
The hypercartiod polar plot supplies the highest directional microphoneDI, In a research lab, the Hypercartiod polar plot DI is ________ dB. The Di reduces in the real world to ________ dB because of the head shadow effect
6
2-3
How does an acoustic signal’s input and output change with 2.5:1 compression ratio
Each time the input signals intensity increases by 2.5 dB SPL, the output signal increases by only 1 dB SPL
How does WDRC help to restore normal loudness growth but linear signal processing cannot?
Linear signal processing adds the same amount of gain to soft, moderate, and loud input signals, white WDRC adds more gain to soft input signals than it does to loud. As a result, when the volume of a linear signal is increased soft signals remain under-amplified and inaudible, moderate signals may be raised so the output is comfortable, but loud signals are then over amplified and perceived as too loud.
In contrast, WDRC adds more gain to soft input signals for improved audibility but slowly reduces gain as the input signal becomes louder. In this way, the amplified signal is squeezed or reshaped into the dynamic range so that soft sounds are audible and are perceived as soft.
moderate sounds are perceived as comfortable and loud sounds are perceived as loud but remain below uncomfortable loudness levels.
Equilization filters
adds LF gain when _____
Vertical telecoil
optimizes collection of signals in a looped room
Horizontal telecoilf
optimizes collection of signals in a telephone
OLC
7:1 CR is applied, _____
WDRC:
1.3:1 CR is applied, ___
Fast acting compression
reduces the chance of discomfort _____ overshoot?
Slow compression release time:
reduces audibility of soft sounds _____
ACG-i:
level detector that _____
Compression shaping channels
allows frequency specific compression of ___
Frequency shaping bands:
allows frequency specific compression of ____
Fast acting compression:
alters the speech envelope ____
AGC-i:
TK input level increases ____
AGC-o:
TK input level decreases ____
which statement about the threshold kneepoint (TK) IS NOT TRUE
Lowering the TX to a softer inout intensity increases the output of the louder input signals
lowering the TK to a softer input intensity will not change the output of the louder input signals
raising the TK to a loader input signal decreases the output of signas below the le
lowering the TK to a softer input intensity increases the output of sigruls below the TK
Lowering the TX to a softer inout intensity increases the output of the louder input signals
Peak-clipping adds extra frequencies to the input signal causing distortion of the output signal. Output limiting compression does not cause distortion. Is this true or false
FALSE- all compression modifies the signal in a way that allows the addition of frequencies which result in distortion. Lower compression ratios are associated with less distortion than higher compression ratios. Linear signal processing does not add distortion
Very soft acoustic signals below the first threshold kneepoint (TK) are attenuated by applying
Expansion
Why is slow-acting compression recommended for patients with reduced cognitive function?
Slow-acting compression is recommended because in last-acting compression, it alters the temporal envelope of the signal and this altering results in a different signal than what was stored in our brain of what the signal should be. Fast acting compression results in us using more working memory and listening effort in order to make sense of this signal that no longen matches what the brain has stored. Therefore, the opposite happens with slow acting compression. it restores the temporal envelope and is easier for those with reduced cognitive function to understand because it better matches what is stored in our brain causing them to use less working memory and less listening effort
Fast Ats significantly change the spectral envelope shape. More working memory is needed to interpret the temporal characteristics of this new speech signal. Patients with lower cognitive abilities have a difficult time “matching” the compressed signal to their memory of the signal
What does the test mic do?
What does the coupler mic do?
What signal is emitted from the T in the box
electromagnetic
What is SPLIT & RSETS?
What is SPLIV & RTSLP? How are they different?
Advantages for each type and disadvantages of each type of earmold material
find the tragus on the mold, antihelix & the aperture
what is input
Intensity of acoustic signal entering the device
what is gain
amount of amp added to input
what is output
Intensity of signal that is delivered into the ear canal
if soft is 50dB and we add 17 at around 3300 Hz, output signal?
output = 67dB
SPL-o-gram
Shows audiometric thresholds in SPL
what is the acoustic horn principle
as a signal arrives to a large horn, channeled down into a small space, the signal is increased (SPL increases)
1: wider opening more sound is collected (more input energy in)
2: longer the horn the more amplification you get (channeling signal into smaller spaces into longer period of time
Increased horn length =
SPL increase energy to the ™
what are frequency response curves
Compare intensity of the input signal to either the added gain or final output of the device
Acoustic benefits associated with hand cupping
Added +5 to 10dB of gain to the input signal in mid to high frequencies
what is amplified in hand cupping
consonant sounds
soft that do not have a lot of energy are collected by the hand and are going into the narrower space to get louder (10-15 dB)
Understand how a carbon microphone works. Specifically, how it converts an acoustic signal to an electrical signal
sound waves come, hit diaphragm and compress it
when it moved in, the carbon balls are pushed together creating pos voltage flow, sound waves go back and forth so when it goes back so does the diaphragm so the balls decompress
compression and decompression of carbon creates a + and - electrical current
sound has positive peaks and negative
these hit d, compress/decompress the carbon balls, and because it created the +/- current flow, the electrical current matched the acoustic sign wave
input signal is blue and comes in it changed and what came out looks the same but it wasn’t sound
transferred acoustic signal to electric signal that looked identical
it is an analog electric signal
Describe the purpose of the receiver
The receiver converts analogous electric signal back to an acoustic signal
Describe the purpose of the vacuum tube amplifier
Vacuum tube amplifier added gain to increase amplitude of the analogous electrical signal
heater element, catho (plate) and by process of the grid opening and closing it could be used to increase/expand the amplitude of the signal
what is a translator
Electrical component used to increase current flow of electronic signals (increases amplitude)
to increase current flow of electric signal (amplifier)
takes analog electric signal and increases its amplitude
amplify the electric signal
Resistor
Electrical component used to add or remove resistance to the flow of electrical current (modifies frequency response or output)
manipulate the signal in some way
electrical component
added to resist flow
maybe we dont want amplitude in all frequencies, maybe we want to add compression to the signal so we use these to change and manipulate the electrical flow of the signal
what are integrated circuit
combine multiple transistors & resistors into a single component
conduct electrical sine signal
transistors
impede the electrical sine signal
resistors
A small resistor control used to modify output signals in early analog HAs
Manipulated output signal, frequency response, amount of gain added & compression ratio
Aka trimmer pot or pot screw
Potentiometer
becomes the helix lock for retention
valley
two goals of impressions
Goal 1: go 2mm beyond
Goal 2: stretch aperture
How easily material flows before it cures; thickness
viscosity
Flows easily with little resistance
Most suitable for devices requiring deep insertion
low viscosity
Thick; more energy to force the material into the ear canal
Provides resistance during flow
stretches aperature
high viscosity
Describe the importance of stretching the aperture
stretching this area for a well fit earmold
if we do not do this we get an uncomfortable earmold because the skin rubs against the plastic
Materials ability to return to its shape after removal
stress relaxation
you can pull on it and it will pop back into shape and won’t change at all
High stress-
Shrinkage over a 7 day period
contraction ratio
distorts as you pull it out of the ear
low stress
After cure hardness
Stability in the impression box
shore value
ideal otoblock placement
2mm beyond second bend
should be larger than entrance of the external canal
foam
should be about the same size as the entrance
cotton
purpose of open jaw impressions
Increases the size of the aperture for a snugger fit
If you have a threshold loss, can you hear soft sounds?
no
Impact of Threshold loss on audibility
i can hear people but I don’t understand them
critical for speech understanding
hf audibility
what supports intelligibility
audibility of hf consonants
The relative contribution of the consonant frequencies to the understanding of speech is shown by the fact that nearly 70% of word recognition is determined by speech energy between
500-2000 Hz
What’s the difference between linear and nonlinear amplification
linear: Adds an equal amount of gain to soft, moderate and loud input levels
nonlinear: increase intensity of soft signals while decreasing intensity of loud signals
what is automatic gain control
Applies different amounts of gain to different input levels
what is dynamic range
range bw softest audible sound and loudest tolerable sound
what is loudness growth
Perception of loudness as stimulus intensity increases
Loudness grows more rapidly for certain listeners than normal listeners with changesin level, reducing the persons dynamic range
abnormal loudness growth
Individuals with threshold loss perceive sound shifting from too soft to too loud more rapidly
abnormal loudness growth
what results in abnormal loudness growth
OHC damage
abnormal growth happens (perception) due to PT not hearing softer sounds because the amplifiers are gone and rapidly the sound goes from soft to too loud
How do modern hearing aids manage frequency specific variations in a person’s dynamic range?
Amplification applies different compression ratios across frequency ranges to shape an output signal into a reduced dynamic range
this is done by manipulating compression in frequency shaping channels
what is frequency resolution
Auditory systems ability to detect discrete frequencies in the cochlea
Describe how frequency resolution changes with SNHL. Why does reduced frequency resolution make it difficult to understand speech in noise?
When frequency resolution is decreased, the primary signal is no longer enhanced making it difficult to differentiate the desired signal (speech) from the undesired signal (noise).
The brain can’t “untangle” the desired speech signal from the undesired noise, so understanding is diminished
what is upward spread of masking
Intense low frequencies mask weaker higher frequencies
what is temporal resolution
AS ability to detect small time related changes in acoustic stimuli over time
Good auditory
what are the auditory processes that support temporal resolution
Gap detection
brief gaps of pauses between syllables, words, sentences, etc (spoon vs soon)
Phonemic duration - differences in duration and order (can vs cant)
temporal ordering - boots vs boost
suprasegmentals - provides meaning (patterns of stress, intonation, rhythm)
temporal envelope vs temporal fine structure
How will audibility of these features impact speech intelligibility?
TFS - Very rapid fine oscillations that provide information on timing within the temporal envelope
Supports detectioin of speech & nonspeech signals in noise
TE - Slow overall change in intensity over time
cues are associated with speech perception in quiet
Describe the benefits spatial hearing supplies
Makes it possible to tell where a sound is coming from in space
determines direction of a sound source
what are interaural timing differences
Amount of time bw sound arriving to one ear to the other ear
One arrives faster than the other
what are interaural level differences
Difference of volume bw two ears
Which frequencies supply the most information on interaural level differences?
High frequencies (>3 kHz) to identify ear to ear head shadow level differences
Which frequencies supply the most information on interaural timing differences?
Low frequencies (<850 Hz) to identify spatial location & sound source
Spatial hearing allows us to
Determine location of a sound source
Unmask sounds otherwise masked by noise
Brain combines and analyzes info arriving from both ears for improved signal detection & identification of speech in noise
Shift our attention and focus on one sound source while ignoring another
Feel connected with the environment
Explain HRTF. What information do these monaural spectral cues supply?
describes the spectral characteristics of sound as measured at the tympanic membrane when the sound source originates in 3D space
Each pinna interacts with incoming sound waves differently, depending on the sound’s source relative to our bodies.
This interaction provides a monaural spectral cue that is helpful in locating sounds that occur above or below and in front or behind us.
Does a discussion of audiometric thresholds sufficiently explain why a patient is experiencing communication difficulties? How could an audiologist supply a patient with a better understanding of their auditory rehabilitation needs during post diagnostic counseling?
Audiograms are not predictive of the activity limitations resulting from a hearing loss
Amplification will not restore any of these functions, in fact, it can make it worse at times
Diagnosing the problem doesn’t help the PT. Providing a comprehensive plan of care that includes amplification does. How you approach the treatment plan will determine if you are a doctor or simply a salesman
Explain the five benefits associated with use of bilateral amplification. Recognize how each benefit supports improved speech intelligibility in quiet or in noise.
1 Allows audibility of ILD, ITD & HRTF signals
2 Binaural loudness summation
Results in PT perceiving greater loudness w/ bilateral devices
In quiet?
Less gain is needed to reach comfortable listening levels
Can fit a PT with less loudness in order for them to hear the words
3 Improved localization
4 Binaural squelch (binaural release of masking)
Signal to noise ratio:
if speech is louder than noise, makes it easier to understand in noise = +
if speech is softer than noise, makes it challenging to understand in noise = -
Leads to improved intelligibility in noise & ability to focus on 1 signal while ignoring others by taking advantage of these differing SNRs
The brainstem uses the differences in speech and noise ITDs/ILDs to enhance focus on speech signals while suppressing noise
Research shows binaural hearing offers an SNR improvement of 2-3 dB
5 Minimizes risk of “unaided ear effect”
6 Suppresses bilateral tinnitus
Explain binaural loudness summation. Its suprathreshold benefits and how this benefit supports hearing aid fitting strategies.
Results in PT perceiving greater loudness w/ bilateral devices
Less gain is needed to reach comfortable listening levels
Can fit a PT with less loudness in order for them to hear the words
@ threshold increase only around 2-3dB
@ suprathreshold increase around 6-8dB
Explain binaural squelch and the benefit this auditory process supplies
ability of the auditory system to combine the information from both ears centrally and segregate the speech from the noise by the differences in sound between both ears.
Leads to improved intelligibility in noise & ability to focus on 1 signal while ignoring others by taking advantage of these differing SNRs
Describe the research findings related to the “unaided ear effect
Research found ~25-33% of individuals with symmetric loss suffered from reduced word recognition scores following 1 year of monaural hearing aid use.
likely due to a central mismatch due to a strong consistent signal received though the aided ear.
The brain pays more attention to the dominant ear overtime, ignoring the ear with the weaker signal
Be prepared to explain each binaural process in a simple way that supports a patient’s understanding of the benefit
amp is less loud and more balance
more natural and better tolerated
easier to hear higher pitches
easier to hear speech in noisy environment
easier to better hear speech in the presence of noise
amp is less loud and more balance
more natural and better tolerated
easier to hear higher pitches
easier to hear speech in noisy environment
easier to better hear speech in the presence of noise
Reduced intelligibility in noise with bilateral amplification
Reported in 5-10% of older PTs with bilateral amp
Progressive age-related atrophy of the corpus callosum reduces speech intelligibility with binaural input
poorer speech recognition with both ears than with the better ear alone
Near normal low frequency hearing typically doesn’t require an amplified signal because the intensity of a “direct signal” is audible without amplification
true
what is feedback
amplified sound that leaks back out and is reamplified
makes squealing sound
What causes a hearing aid to feedback?
HA isn’t snug enough in the ear, output leaks out side of HA and the amp signal is picked up by the HA and creates a feedback loop
What causes patient complaints of occlusion?
increased perception of your own voice when something blocks the ear canals
can occur with ha that has dome that closes off ear or HA with earmold with not large enough vent size
What happens when a microphone gets closer to a receiver?
closer microphone and receiver are together the less output you can get because it drives feedback
Describe signal changes resulting from the pinna effect
Pinna naturally adds gain & provides valuable spatial awareness cues
The pinna effect boosts the intensity of HF signals (specifically 3k Hz region) by ~7 dB SPL
How will microphone placement impact the intensity of the output signal arriving to the TM?
The combination of the pinna effect & microphone location will increase the output signal and decrease wind noise
CIC & IIC get pinna benefits the most
How will the canal volume b/w the receiver and TM impact the intensity of the output signal arriving to the TM?
Sound bore proximity to ™ increases device output
The output arriving to the TM increases by ~6 dB SPL when the volume b/w the TM and receiver reduces
The combination of a deep microphone placement and deeply fit devices increases output by ~13 dB SPL
What does an IP rating indicate? What does a 6/8 IP rating suggest?
what we look at in BTE to decide is the device dust and moisture resistance
IP68
Dust light
Protects against long periods of liquid immersion
BTE fitting range
all degrees of HL
no pinna effect
output can modify based on acoustic parameters
what are the acoustic parameter
venting
damping
sound bore
slim tube fitting range (RITA)
mild to mod
25-55
no PE
HF output is limited by vent size and tube soundbore
RIC fitting range
minimal to severe
-10 to 90
no PE
output dependent on receiver size and earmold style
ITE fitting range
WNL in the lows (250-1) to severe (>1)
-10 to 70
minimal OE
output limited by receiver & concha size & depth in the canal
ITC fitting range
around 20 dB (WNL) to 500 Hz then mod severe
20-70
minimal PE
output limited the same as ITE
CIC fitting range
mild to mod severe
25-70
has PE
output is better due to depth of mic and receivers proximity to TM
IIC
mild to mod severe
best for LF HL
25-70
has PE
output increases up to 7dB if mic depth allows PE & up to 6 dB due to proximity to tm
what is a cros
Contralateral routing of sound
Single sided deafness - one ear is normal and poorer ear is unaidable
What is a bicros
Bilateral contralateral routing of sound
Bilateral asymmetric HL - one ear has threshold loss & poorer ear is unaidable
for cros or bicros when the PT wears the devices will they get binaural hearing?
No because they still only have one ear and will not have spatial awareness
what is ampcros
For asymmetric HL
bad and better ear
bad ear is not good for hearing aids (poor discrimination etc.) but not unaidable
fits two HA’s on the PT’s ear but in the two is a transmitter so you amplify both ears but then also routing the poor ear over to the other ear for extra boost of understanding
adv and disadv of vinyl and when would it be used
used for infants, firm ears, high gain devices, facial flex issues, older adults w/ dexterity issues
adv- snug fit for high gain, easily modified, inexpensive to remake
dis - shrinks, hardens, discolors, requires replacement every 6-12 mos
adv and disadv of silicone and when would it be used
used for peds, high gain devices, allergies, facial flex problems
adv - durable, doesn’t shrink, hypoallergenic for most
dis - sticky grip can cause blisters, hard to modify, costs more
adv and disadv of lucite/acrylic and when would it be used
used for adults, mild to severe losses, floppy or soft pinnas
adv - no shrinkage, durable, discreet, easily modified, smooth surface for insertion and removal
disadv - increased risk of feedback w/ movement, potential injury if hit, harder to move beyond narrow/tortuous areas
Understand the importance of the acoustic seal
Understand the importance of the acoustic seal
a tight acoustic seal reduces feedback
full shell fitting range
used for higher output in severe to profound
70- >90
maximized retention & acoustic seal
skeleton mold fitting range
mild to sev
25-90
subtle
maximized retention & acoustic seal
canal mold fitting range
mild to sev
25-90
retention needs 2mm beyond 2nd bend
lowest degree of acoustic seal
canal lock fitting rang
mild to sev
25-90
Adds projection into concha for improved retention by antitragus
half shell fitting range
Mild to severe HL
-25 to 90 dB HL
Fills ½ of concha to improve retention & ease insert
Describe the cause of the occlusion effect (OE) and associated patient complaints. Describe the degree of LF threshold loss associated with OE concerns
Describe two management techniques used to reduce OE
Skull transduces LF energy generated by your own voice and the signal becomes trapped in the ear canal
Increased perception of one’s own voice when something is blocking the ear canal
Common complaint occurs when LF threshold is better than 50 dB
Causes
Dome closes off the ear
Mold with too small of a vent
Insufficient venting and/or insufficient canal length
Vent size & stabalizing the device in the bony canal are two management techniques to reduce OE
fitting range for open domes
</= 20 dB up to 1.5 kHz
fitting range for closed dome
20-29 dB at 500 Hz
thresholds better than 40 below 1khz
fitting range for power dome
30-39 at 500 Hz
thresholds better than 40 below 1kHz
Name the primary frequency range impacted by the vent effect
primarily impacts frequencies below 1.5k Hz
“direct signal“ and “amplified signal”
Direct Signal
Refers to unamplified signal arriving to the TM
Allows PT to hear LF signals & environmental sound naturally when LF thresholds are better than 40dB
Direct Signal
Refers to unamplified signal arriving to the TM
Allows PT to hear LF signals & environmental sound naturally when LF thresholds are better than 40dB
the two combine being perceived as one signal arriving to the ™
Amplified signal is 20 dB SPL louder than the natural direct signal
audibility of the amplified signal dominates
If intensity of direct signal is louder than the amplified signal
direct signal will mask the amplified signal
Describe the impact of standing waves occurring inside a vent
The final output and frequency response arriving to the TM may be unexpectedly altered when the direct signal moving inward is in-phase or out-of-phase with the amplified sound moving outward
in phase double the intensity
out of phase signals cancel portions out
what is vent size needed for 50-60 dB at 500
.5 to no vent
what is vent size needed for 40-49 dB at 500
1-2 mm
what is vent size needed for 30-39dB at 500
2-3 mm or power
what is vent size needed for 20-29 dB at 500
3-3.5mm or closed
what is vent size needed for </= 20 dB at 500
open
A change to the internal diameter will change the device output and frequency response
true
smaller diameter of inner tube
more attenuation in high frequency signals
Outer wall thickness increases to
reduce tube vibration
This reduces feedback in high gain devices
tube lock is used
w/ silicone molds
brass grommet for friction in mold
libby horn is used
as amp signal goes thru tube to horn you get a boost of amp in the high frequencies
increase output of highs by about 6dB
PT not ready for new ha and need a little more output in highs, create a mold with this to help some
Continuous flow adapter (CFA) is used
for small canals that cannot accommodate standard tube sizes
Maintains internal diameter of sound bore bw a BTE & ear mold providing a continuous inner diameter & unimpeded flow of amplified sound
Dry Tube is used
to reduce moisture buildup issues in tubinh
Describe the impact of hardening tubes on the device’s frequency response output signal
Length shrinks
Displaces mold causing increased feedback
Hardens
HF gain reduces as inner diameter shrinks
thin cement is used with
lucite/acrylic
vinyl cement is used with
vinyl
If HL 50dB or more in low frequencies you have to have a mold
true
Describe the piezoelectric effect and the limits of piezoelectric microphones
twisting, compressing or distorting a thin electrified crystal creates a +/- electrical voltage required to make an analog electrical signal
This salt crystal replaced carbon balls in the mic for early electric HA’s
*this mic was short because it was affected by humidity and temps >110 deg F
Compare & contrast Electret microphones and MEMS Microphone
electret - years of humidity, moisture, dirt, debris etc. degrades the mic & reduces sensitivity of the mic
MEMS - Stability: more stable
Silicone doesn’t absorb moisture
Has decreased battery drain in the device due to space being smaller bw diaphragm and backplate
Define microphone sensitivity
What input frequency range can a microphone collect?
ability of the mic to pick up sounds
Sensitive collection from <100 Hz to as high as 15000 Hz
Differentiate the causes of acoustic noise from the causes of electrical noise.
Acoustic - gasses & air flowing around us goes into sound port and moves the diaphragm
Electrical - comes from circuits in HA that are after the mic in ha
What is an acceptable intensity of a mics internal noise floor?
~ 25 dB SPL
What is the dynamic range of an analog hearing aid microphone? Of a digital hearing aid microphone?
Analog mic = 115 dB SPL (can collect input signals up to this before input distortion
Digital mic = ONLY 96 dB SPL (anything > results in input distortion)
Define front-end distortion and its cause. Describe why digital HA’s have a lower mic dynamic range. Describe how front-end distortion is managed in digital hearing aids.
mic has a dynamic range and FED happens when the input signal exceeds this
Distortion in analog - input >/= 115 dB SPL (anything louder sounds distorted)
Distortion in digital mic happens with input >/= 96 dB SPL (0-96, anything louder sounds distorted)
Shift mic dynamic range by lifting the 96 higher to collect more loud sounds but does so by sacrificing soft
If we shift the range of mic in loud situations you can collect loud sounds without peak clipping but can’t collect soft
If we shift range in soft situations you can collect soft sounds but not loud ones anymore
Move form 16 bit to 19 bit HA in other digital devices allowing 108 dB dynamic range
why digital hearing aids have a lower microphone dynamic range
due to analog to digital converter (ADC)
Used to transduce an analog electric signal to a digital signal
16 bit ADC supplies a 0-96 dB dynamic range and louder sounds are peak clipped
Describe the impact of microphone distance on SNR
SNR becomes poorer as distance bw mic and desired signal increases
Further it is from the mic harder it is to hear over the noise because noise becomes louder - reduced SNR
Closer to the mic the louder it is over the noise - Improved SNR
What is the ideal distance for sound collection?
Ideal mic sensitivity w/in 6 ft
Which frequency range is associated with undesired background noise?
<1500 dB
DI 0.0
omniDI
DI 4.8
cartiod
DI 5.7
Supercartiod
DI 6
hypercartiod
How does a directional mic use in-phase and out-of-phase signals to create a polar plot null
If acoustic signals arrive at the same time in phase, they move the diaphragm creating an analog electric signal
In phase = analog electric signal
If acoustic signals arrive at different times and out of phase they STOP diaphragm movement not creating an analog electric signal
Out-of-phase = STOP diaphragm and no analog electrical signal
function of a directional microphone so you can explain how signals arriving from behind a patient are attenuated
Signals arriving from behind reach the two ports at different times creating an external time delay.
The delay causes out of phase signals that doesn’t allow the sounds to pass through and push on the diaphragm. This causes the sounds from behind to be attenuated
Define directional mic roll-off. What causes it and how is it managed?
Happens when directional mics are turned on
because LF are broad and wide they are more likely to arrive out of phase causing them to be lost
results in 6dB/octabe LF roll off causing a reduction in volume
managed by equalization filters that add LF energy to replace the attenuated output signal
Describe the importance of electret microphone matching and calibration
stop working when they fall out of calibration
has to be positioned parallel to the floor to keep the external time delay
If difference is greater than 1dB you lose directionality
What causes reduced microphone drift?
Microphones that fall out of calibration over time
Occures due to:
High temps reducing mic sensitivity
Moisture causes damage to diaphragm tension ring
cerumen/debris clog mic port
When something plug the port, the sound coming in from the back doesn’t reach the diaphragm which doesn’t cause an out of phase signal which means no directionality occurs
Describe the principle of induction
Telecoil uses this principle to transduce an electromagnetic signal to an analog electrical signal
Mic is turned off (NO ACOUSTIC SIGNAL)
takes electromagnetic signals and transduces it into an electrical sine wave
Electromagnetic signals move a magnet with a copper coil
This movement bw copper coil & magnet turns electromagnetic signals to a +/-analog electrical signal without adding an additional power source
Benefits of telecoil
inexpensive
no external power source needed because magnetic field generates its own
no telephone feedback because mic is off
improved SNR on phone or in loop room
pick-up room loop signals
vertical telecoil
pick-up telephone signals
horizontal telecoil
Engages AFTER volume control
Output compression (AGC-o)
Engages BEFORE volume control
Input compression (AGC-i)
what is threshold kneepoint
Changing from 1 compression ratio to another
> /= 85dB SPL
Used to limit output of a HA so it doesn’t exceed the individual’s loudness discomfort levels & to maximize listening comfort
high TK
</= 50dB SPL
Used to improve audibility of softer components of speech and/or
threshold that when amp reaches it compressor kicks in
low TK
time it takes for level detector to identify input is loud enough to turn on and compress the signal
attack time
shorter duration of overshoot & shorter period of time individual hears sounds louder than desired
faster AT
Period of overamplification as input SPL increases above TK
Output signal overshoots targeted SPL
slow AT
Amount of time for compressor level detector that the intensity fell below TK and needs to release the compression
release tie
sudden loud sounds
fast attack
best for conversational speech to maintain the shape of the spectral envelope
slow attack
brief, intense sounds (door slamming)
fast release time
longer intense sounds (such as a raised voice)
long release time to maintain a comfortable output level during brief gaps of silence
How much compression to apply
compression ratio
2.5:1 CR
For every 10 increase in input intensity, output increases by 4
Presence of frequency components in HA’s output that were not present in the input signal
harmonic distortion
shifting tk down
adds signal below TK
increasing tk up
removes signal below TK
Differentiate AGC-o from AGC-i
I - WDRC
b/w 20-50 dB SPL
Manages incoming sound
input compression
O- OLC
>/= 80dB SPL
manages loud sounds at output level
good for those with lower cognitive capacity because temporal envelope is not compressed and looks more like what our brain stored of the sounds; while someone speaks, amp of temporal envelope shape is closer to auditory memory of signal for longer periods of time
slow compression
gets compressed, work harder to match sound is sort of a memory to understand and causes more listening effort resulting; doesn’t take a lot of volume for it to compress & envelope doesn’t look like memory of it working harder to understand what was said
fast compression
needed when someone complains about some of soft sounds they do not need to hear are too loud
expansion
Lower output with very very soft sounds
really low CR (<.9:1)
0-20dB input
expansion
needed to get soft sounds louder & expand dynamic range
manages incoming signal
Engages BEFORE volume control
input compression
activates at the pre-amplifier when TK is low input level & signal is louder than this (bw 20-50)
low TK
low CR (1.1:1-4:1)
WDRC
need this to protect the ear
AGC-o
loud sounds
engages after vc
high tk (>80)
high CR (>/= 5:1)
output compression
allows frequency specific compression of
Frequency range is defined by the frequencies included in range based on dynamic range of loss
Frequency response is a series of volume controls called bands/handles
frequency shaping bands
shape compression characteristics into individuals dynamic range to restore normal loudness growth
use these channels to squeeze gain for soft & loud into the persons dynamic range
compression shaping channels
as input increases amount of gain applied is reduced
compression
as input increases amount of gain applied is rapidly increasing too
expansion
Very soft acoustic signals below the first threshold kneepoint (TK) are attenuated by applying
expansion
Explain the process of converting:
An analog electrical signal to a digital signal
Include an explanation sampling rate, Nyquist frequency, quantization, quantization error, and processing speed
You should have an understanding of how each of these factors impacts the output signal
A digital signal back to electrical
Sound comes into mic (either in or out of phase; in goes on out is nulled), MEMS or ECM mic transduces acoustic signal by compressing and decompressing the diaphragm and the backplate creating the + and - analog electrical signal
NOW, sound leaves mic and goes through a bunch of circuits that either compress or amplify the signal
Signal is either amplified (transistor) or compressed (resistor) and is either routed to AGC-i or AGC-o
Then sound goes to the analog to digital converter to go from an analog electric signal to a digital signal
ADC - when converting an analog signal to a digital signal the signal is sampled at discrete intervals (sampling rate)
Digital signal now has numbers associated w/ it and moves to DSP
Moves to Receiver
converts amplified electric signal back to an acoustic signal
*acoustic signal, transduced to electric at mic, amp at different levels, electric changed to digital signal, algorithms added, digital converter back to electric analog signal that is amp, now to receiver to transduce back to acoustic signal
number of times per second an analog signal is sampled to create a digital signal - how we capture frequency information
sampling rate
more snapshots taken- the more accurate the original continuous wave is represented due to more sampling points
high sampling rate
Nyquist theorem
Half of the sampling rate
If sampling ate is 20,000 Hz, nyquist frequency is 10,000 Hz
more possible vertical amplitude values & more precisely the exact amplitude of a given sample can be recorded
higher bit depth
Higher quantization = more fluid sample
Lower = more choppy sample
Rounds each snapshot to nearest fixed level
quantization (bit resolution)
The difference between the original acoustic signal and the transduced digital signal
quantization error
1 bit = front end DR by _____ dB
6
16-bit digital word = DR of 96 dB
18-bit digital word = DR of 108 dB
quantization error
Noise floor in the hearing aid is noise created in the circuit due to
quantization error
Explain front end limitations associated with 16-bit processing, and how this impacts microphone sensitivities. How are these limitations resolved?
16-bit processing has a dynamic range of 96 dB - it can collect sounds up to this but past this the sound becomes distorted
We get this dynamic range because each digital bit increase the front end dynamic range by 6dB
Every bit only has 6 dB of dynamic range
Solution: the dynamic range shifts by lifting the 96 dB higher to collect more sound sounds but it does this by sacrificing soft sounds
The function of auditory filters (critical bandwidths) in the cochlea (included number and size of filters
There are 25 bands that do this
Narrow LF bandwidths - only 160 Hz wide)
Wide HF bandwidths - up to 2.5 kHz wide)
Cochlea is a series of overlapping band pass filters (frequencies that are grouped together because they are close together on the cochela) that allow certain regions on it to stimulate to a specific frequency region while ignoring frequencies outside of the band
The effect of low frequency masking on the damaged cochlea.
Broadening of the filters is mainly on the LF side, leading to the increasing of LF masking. So the LF noise masks the region it normally would but also spills over to the overlapping HF bands around it making it more difficult for the PT to use the cues to understand the speech over the noise
w/ HL, the curve is broader and noise can easily affect perception of the desired signal
Frequencies impacted by noise
Noise energy peaks around 250 Hz but upward spread of masking impacts audibility up to about 1500 Hz
what are methods of sound cleaning technology used in the spatial domain
automatic mic switching (fixed, adaptive, beamforming)
ONLY thing we can do in digital ha tech to improve speech intelligibility is
enable direction mics
fixed vs adaptive directional mic
fixed - one polar plot
adaptive - multiple polar plots
superior when only a few noise sources are present
adaptive
superior in the presence of multiple noise sources
fixed
what is beamforming mic
Has a very narrow beamwidth (only azimuths of + 25°, + 35°, + 50°
Monitors overall intensity of environment
When signal is <55 SPL widest beamwidth activates
As environmental intensity increases (>75 SPL) smallest beamwidth is activated
what are the methods of sound cleaning technology used in the temporal domain
modulation rate vs depth
digital noise reduction
modulation rate vs depth
Speech = slow rate
Noise = fast rate
depth
Speech = highly variable
Noise = steady over time
fast modulation rate & depth is stable over time =
noise
Digital noise reduction: describe the attenuated signal
Describe the benefits and limitations of digital noise reduction
acts on steady state noises (idling engine, hair dryer, vacuum etc)
only acts on fast mod rate & low mod depth
doesn’t improve speech intelligibility
Can improve listening comfort, reduce listening effort, reduce cognitive load
what are methods of sound cleaning technology in the spectral domain
HA here looks at a signal’s frequency to control the signal
Theory is that If noise is below 1.5 and then understanding speech comes from mid and high frequencies reducing the output in low frequencies will improve speech intelligibility in noise
if we attenuate and reduce lf amp and leave in hf, then we will probably improve speech intelligibility
not a lot of improvement by doing this
some noise is hf and not lf
ex: listen to music and speech is disruptive (talking in background) when you reduce lf then you take bass away but still hear person talking
Explain Weiner filter function and limitations
Spectral subtraction approach, measures short term noise spectrum during gaps in speech
Works good on steady noise but not fluctuating so it is not effective in real time noisy situations
Reads between the lines in desired signal and takes out background noise found in the modulations of speech
what are 3 methods to reduce external feedback
adaptive digital feedback suppression (DFS)
- reduce external feedback loop
- digital notch filtering
- digital feedback cancellation
describe reducing external feedback loop for ADFS
Increase snugness of mold to reduce size of slit leaks
Or decrease vent size to stop feedback path
Limitation: both increases OE
describe digital notch filtering in ADFS
Removes frequencies around the noise - reduces gain around 2-4 kHz where feedback occurs
Limitation 35% of intelligibility comes from this range alone so you stop feedback but stopped audibility of important speech sounds so reduced speech intelligibility
describe digital feedback cancellation in ADFS
When HA detects feedback (identified due to steady state noise bw 2-4 kHz) an algorithm creates an out of phase clone of the signal (duplicate of the feedback) and this causes the clone to be subtracted from the amp path and in turn attenuates the feedback
Describe the limitations of each feedback reduction method (DFS)
Algorithms can cause brief feedback until the out of phase tone can activate
Any sustained tone can start an algorithm (whistle, violin, etc.)
Audible beep when an external signal is mistaken for feedback - entrainment
Feedback cancellation can distort or attenuate parts of speech
Faster battery drainage and life
what are 3 types of frequency lowering
linear frequency transposition
nonlinear frequency compression
spectral envelope warping
when is freuqency lowering used
Only works from high to low
Best for steeply sloping HF HL
tries to improve HF audibility by shifting it down to LF
used with someone that has residual hearing and loss in HF makes it impossible to reach hf amplification (cannot get them to ever hear these sounds again)
what is linear frequency transposition
CUT AND PASTE into LF, takes the highs and shoves it into the lows
Improves HF audibility by moving HF band one octave down to LF region
what is nonlinear frequency compression
HF range is compressed into a LF range; squishes it down into the audible region
maintain tonotopic order more
what is spectral envelope warping
Leave the HF where it is but also COPY and PASTE into LF
Keeping a portion in HF but also transposing portion down to LF range
Describe digital wind noise reduction methods
Wind can increase output by 20-25 dB
Wind turbulence only affects one diaphragm so the HA talk to each ither and take the signal on the opposite side and transmit it and overlay it so it seems like the noise went away and improves snr in that environment
benefits of wireless binaural processing technology
wind noise management
dual phone
volume control
program
WDRC
how does wireless binaural processing support improved awareness of ILD in HA’s
As it arrives to second ear it is lower and wdr adds more gain on that same side so the ILD are gone so you cannot hear well in noise so binaural wireless the one ha tells that ha to not add as much wdrc
One ear will tell the other ear to not add as much wdrc so it doesn’t lose ILD
automatically senses if it is noisy or not & decides if omni or directionality is needed
automatic mic switching
adaptive vs fixed directional mic
Adaptive: null repositions itself (steer) to the loudest signal behind the person
Uses all the polar plots
Fixed: Picks a polar plot and stays using this position
Which is better for high noise environments?
Fixed is better for more noises because it attenuates all the signals
Adaptive is better for a few noise sources because it is not smart enough to determine which ones to attenuate; it doesn’t know how to function
Narrowband
Directional mic focuses in on a very narrow field (25-30 deg)
Good for talking with one person at a time and block out as much sound as possible
Can be to front, side (if driver in car), in back if you want to hear someone from behind
beamforming
only circuit in HA that will improve this in noise
directional mic in the spatial domain
After DSP, digital data stream is converted back to the analog electrical signal by
Converting from digital signal to analog electrical signal and goes directly to the receiver (less distortion, less noise, cleaner signal)
Or digital signal is converted to analog electrical signal that then gets amplified by an AGC-o (output amplifier to boost or add output compression) before entering the receiver
Sets guidelines & standards for all diagnostic equipment and for instruments etc.
ANSI
Compare results to standards of manufacturers and ANSI
SPEC sheet
Acoustic chamber
Reduces reflections
Low ambient room noise
Calibrated sound sourcc
test box
Emit measurement signal
speakers
Calibrates SPL output from speakers
reference micc
Measurement mic collecting output from the HA
coupler mic
Simulates size of the canal
Custom products - ITE & ITC
Uses fun tak to attach HA to it
HA-1 coupler
Traditional BTE w/ earhook
Attach directly to hook
HA-2 Coupler
Only for verifit 2
Used for CIC, IIC, RIC
Smaller cavity volume mimics the deeper placement of the device
Realistic picture of frequency response in >HFs
ANSI requires use of same coupler used by manufacturer
.4cc wideband coupler
Better simulates characteristics of the ear canal
Avg canal has HF SPL that is higher than the LF SPL
zwislocki coupler
This shows output for a 90dB SPL input signal (MPO)
Loudest possible output point device can produce for a 90 dB input signal & represents a single frequency
OSPL 90- Output SPL @ 90 dB
Calculates the average OSPL 90 output for 1, 1.6 and 2.5 kHz
*3 triangles at the bottom
Useful to tell what the max output of HA is and whether it will get too loud for the PT
HFA OSPL 90- HF Avg output SPL @ 90dB
Shows avg gain for 1, 1.6, 2.5 for a 50 dB signal
Estimates the max gain available at different frequencies when an avg input signal is amplified
HFA FOG - HF avg full-on gain
Measures the internal noise of the HA
Typically 25- 30 dB SPL is acceptable
equivalent input noise (EIN)
Measures signal distortion
Determines if output signal contains harmonic frequencies that were not present in the input signal
total harmonic distortion
Should be below 5-10%
High DL are indicators the device is close to failing (most likely the receiver) & needs sent in for a repair
% THD
3 steps for looking at ansi measurements
step one: look at manufacturer value on spec sheet
step two: look at summary of tolerances
step 3: look at data point; is it in or out of spec
what is emitted from the T in the test box
electromagnetic signal
what is SPLITS & RSETS
what is SPLIV & RTSLP
How are they different
SPLITS - measured output response of the HA w/ electromagnetic signal
RSETS - calculates the difference between the mic output and the telecoil output (SPLITS)
+ RSETS = telecoil output it louder than the mic output
PT will turn HA down when telecoil is on
- RSETS = telecoil ouput is softer than the mic output
PT will turn HA up when telecoil is on
SPLIV
SPL in a vertical magnetic field
Looped environment
RSLS - relative simulation for loop sensitivity
SPLIV - output of telecoil
RTLS - difference bw mic and telecoil
what if SPLITS is larger than SPLIV
telecoil is more sensitive in the horizontal position
what if the SPLIV is higher than the SPLITs
telecoil is more sensitive in the vertal position
output of the telecoil
SPLITS
difference bw mic and telecoil output
RSETS
+ RSETS
tcoil output is louder than the mic output
- RSETS
telecoil output is softer than the mic output
first limitation to frequency response of HA
sampling rate
ADC sampling rate determines the highest frequency a device can produce
dynamic range of mic is impacted by
bit rate of an ADC converter
Describe the purpose and function of a receiver
converts amplified electric signal back to an acoustic signal
armature (flexible strip of metal balanced between two magnets like a diving board) is magnetized as the electrical current flows through the coil and it moves up in the positive direction towards that magnetic and down to the negative magnetic mimicking the electrical sine wave which is attached to the diaphragm above it and as the armature moves so does the diaphragm and this diaphragm movement causes the push and pull of the air creating an acoustic signal
How are receivers designed differently to achieve the greatest high frequency output for severe hearing losses
Receiver sizes determine the final HF output
HF signals need rapid diaphragm movements
This is done w/ smaller receiver because it makes the diaphragm smaller and stiffer
Smaller contemporary receivers are capable of
higher frequency responses
as high as 10-12 kHz
Trying to achieve HF in large receivers compromises LF gain that is needed for severe HL
true
what are dual receivers (two receiver system)
Output from both receivers added together when it reaches the ™
One is for LF
One is for HF
2 receiver system that sums both receivers before reaching the ™ where one is optimized for LF and the other HF which reduces battery drain, gives a good EHF output w/out compromising a lot in LF and minimizes saturation distortion potential
Benefit:
Extended HF bandwidth in moderately high output receiver without compromising the LF
This happens when the receiver output is reached resulting in peak clipping or the receiver having a higher voltage and battery drain
HA output range has been exceeded leading to distortion due to peak clipping
Receiver Saturation
Caused by a dislodged receiver where the vibration goes back into the mic adding extra frequencies to the input signal leading to distortion
shock damage
Can block the receiver diaphragm and cause reduced output of the receiver
cerumen/moisture/debris
Define soundbore path
A column of air that sound waves pass through leaving a receiver that arrives to the ™
Final output & bandwidth (frequency response) of device changes based on a sound bore paths acoustic parameters/physical properties
true
RIC devices also have a sound bore path
false
What frequency range is impacted by changes to the sound bore
Sound bore flaring
HF output is changed based on the side of the flare
Increases HF depending on length and size of the flare
Thin tube
Making the tube thinner decrease the HF output & shifts the peak resonance down to lower frequencies (1000 hz down to 800)
Describe the impact of standing waves within the soundbore on the final output signal
This happens when reflection in the sound bore causes the wave to overlap with itself
This causes two waves of the same frequency to either be in phase creating a single sound that is louder or two identical waves 180 deg out of phase and cancel each other out
Smaller internal diameters of a thin tube cause reduction in
HFO & shifts tube resonance down to around 800 Hz
libby horn
internal horn shape
increases HFO
what is an issue with libby horn
benefit depends on maintaining length of flare canal
cannot just be a flare has to have a certain length
cannot always get diameter or length to fit into individual horn
Depth of soundbore in canal
Increased length = Increased SPL for ALL frequencies
Describe the purpose and limitation of a damper
Acoustic resistor
Physical barrier to reduce sound
Smooths the resonances in the final frequency response
Does so by attenuating sounds slightly to smooth the peaks
Sound waves lose energy as they travel through the damped material
Higher ohm = more resistance
true
if a damper is replaced, any color can be put in
false
must be the same color
Course
Used to modify silicone
Need to use at least 25k to 20k RPM for best results
blue stone
For grinding acrylic or vinyl
Smaller one is for smaller areas
white stones
Removes large amounts of material fast
low grit
Smooths and restores shine
Only used w/ lucite/acrylic molds
fine grit
Removes less material and smoother finish
medium grit
Final edge smoothing
nail file
pros of rechargeable batteries vs disposable
Rechargeable
Convenience
No small batteries
No keeping track of batteries
More eco friendly
No buying batteries
Disposable
Convenience
Smaller HA size
Lower HA cost
Better reliability
what is the purpose of a listening check
subjective eval of HA, will only pick up significant device malfunction, should perform electroacoustic measure to compare to manufacturer specifications
Listening for unusual distorted bad sounds
Is it linear or nonlinear
Is it directional or omni
Hold the device _____ from sound source
~12”
linear ha
If there is no change in amplitude from soft to loud
nonlinear ha
Amplitude is louder for soft and softer for loud
directional mic
If you turn HA 360 deg and sound attenuates behind HA
omnidirectional
If you turn HA 360 deg and it doesn’t attenuate from behind
what are the 6 ling sounds? why are these used
“ah”, “ee”, “oo”, “sh”, “ss”, “mm”
Used to test across a frequency range
what is SBAR communication method to relay a call for action to another healthcare provider
Get to the point asap
Situation: brief statement of the problem you are facing
Background: follow up with pertinent case details
Assessment: statement of concern
Recommendation/request: what are you asking of them
what is patient centered care
Active involvement from PTs and their families to design new care models in decision making about their options for treatment
Takes time & dont get reimbursed but we can for other services - need to bill & code effectively to be reimbursed for services we do provide
when will medicare reimburse
Evaluation of the cause of disorders of hearing, tinnitus, or balance;
Evaluation of suspected change in hearing, tinnitus, or balance;
Determination of the effect of medication, surgery, or other treatment;
Reevaluation to follow up regarding changes in hearing, tinnitus, or balance that may be caused by established diagnoses that place the patient at probable risk for a change in status, including but not limited to otosclerosis, atelectatic tympanic membrane, tympanosclerosis, cholesteatoma, resolving middle ear infection, Meniére’s disease, sudden idiopathic sensorineural hearing loss, autoimmune inner ear disease, acoustic neuroma, demyelinating diseases, ototoxicity secondary to medications, or genetic vascular and viral conditions;
Failure of a screening test (although the screening test is not covered);
Diagnostic analysis of cochlear or brainstem implant and programming;
Audiologic diagnostic tests before and periodically after implantation of auditory prosthetic devices.
will medicare reimburse?
Evaluation of the cause of disorders of hearing, tinnitus, or balance;
Evaluation of suspected change in hearing, tinnitus, or balance;
Determination of the effect of medication, surgery, or other treatment;
Reevaluation to follow up regarding changes in hearing, tinnitus, or balance that may be caused by established diagnoses that place the patient at probable risk for a change in status, including but not limited to otosclerosis, atelectatic tympanic membrane, tympanosclerosis, cholesteatoma, resolving middle ear infection, Meniére’s disease, sudden idiopathic sensorineural hearing loss, autoimmune inner ear disease, acoustic neuroma, demyelinating diseases, ototoxicity secondary to medications, or genetic vascular and viral conditions;
Failure of a screening test (although the screening test is not covered);
Diagnostic analysis of cochlear or brainstem implant and programming;
Audiologic diagnostic tests before and periodically after implantation of auditory prosthetic devices.
yes
when will medicare not reimburse
When the auditory/balance status is already known
When the reason for the hearing assessment is unrelated to hearing aids, or examinations for the purpose of prescribing, fitting, or modifying hearing aids
will medicare reimburse?
When the auditory/balance status is already known
When the reason for the hearing assessment is unrelated to hearing aids, or examinations for the purpose of prescribing, fitting, or modifying hearing aids
no
what are CPT codes
codes used to describe primary diagnostic procedures done
what are ICD-10 Codes
codes used to classify a diagnosis or symptom
what are hcpcs
codes describing services & supplies that are not defined or outlined in CPT codes
can you code for doing a hearing check and repair? if so what are the codes
yes
**look up
