Frequency Shifting for Patients with High Frequency Hearing Loss Jack Ho, Joseph Yuen, Nate Werbekes, Kuya Takami Advisor: Thomas Yen, Ph. D. Design Criteria. Abstract. find maximum and minimum frequencies patient can hear gets entire range of what normal humans can hear
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Jack Ho, Joseph Yuen, Nate Werbekes, Kuya Takami
Advisor: Thomas Yen,Ph. D.
Our group uses digital signal processing chips from Texas Instruments to create an improved hearing device for patients with sensorineural hearing loss (also known as high frequency hearing loss). Our device will first ascertain the maximum and minimum frequency that the individual is capable of hearing. Once these values are known, the device will compress and shift all the sounds a typical human can hear into the impaired range of our patient. This semester we planned to build the hardware to control input and output, and learn the development kit which will program the actual DSP chip. The DSP chip we have selected, the C5509, is designed for use with audio signals and has a built in ADC. We decided to use LabVIEW in place of the development kit to construct a working model of the hardware.
Another method is block-based time domain pitch shifting technique (transposition).
(a) shows input signal split into blocks;
(b) for a down pitch shift, input blocks are truncated by an integer number of periods to create a signal of shorter duration;
(c) for an up pitch shift, input blocks are overlapped and added to increase signal duration during the calculation of the inverse transfer function.
This method could move the frequency without changing the tempo of a signal. Because this method does not require calculation of frequency domain signal using Fourier Transform, this method can modify a signal faster.
Time shifting of overlapping blocks; (a) depicts an input signal split into 3 overlapping blocks; (b) blocks are shifted forward in time to increase signal duration; (c) blocks are shifted back in time to decrease signal duration (http://www56.homepage.villanova.edu/scott.sawyer/fpga/II_freq_domain.htm)
One way to perform this operation is resampling. By performing a Fourier transform function the signal is converted from the time domain to the frequency domain. A signal in the frequency domain could be shifted or compressed in the frequency domain so that the all the speech frequency range could fit within the frequency range of a person with sensorineural hearing loss. Shifting a signal in the frequency domain would not cause expansion in the time domain; however, it would result in an overlap of certain frequencies. On the other hand, compressing within the frequency domain would not result in an overlap of frequencies; it would lead to an expansion of the signal in the time domain.
A phase vocoder method was used for the modification of the frequency in frequency domain. The operation of the vocoder is dividing the original audio signal to shorter frame, and performs manipulation in the frequency domain using Short Time Fourier Transform (STFT), which is the discrete Fourier transform of a short, overlapping and smoothly windowed block of samples.
Signals in frequency domain frequencies are modified by changing the amplitude and phase of frequency. Then the signal is resynthesized by converting the signal back to the time domain using the inverse STFT, which is the inverse Fourier transform on each chunk and adding the resulting waveform chunks.
Conductive Hearing Loss
-in middle or outer ear
-wax, infection, or foreign object in passageway
-dampens all frequencies
Sensorineural Hearing Loss
-damage to inner ear or auditory nerve
-plethora of causes, some are more common
*birth defects, loud noises, aging
-loss of certain frequency bands (usually high)
-no amount of amplification will help
Typical Hearing Aid
-works by amplifying all sounds
-unnecessary amplification of most frequency bands
-doesn’t help those with sensorineural hearing loss
-works by analyzing sound and directly stimulating the appropriate region of the cochlea
-limited by number of stimulators
-typically poor overall quality of “sound”
Hearing test for frequency: A LabView program runs a “quick” frequency sweep from about 10 to 20k Hz, where the user pushes a button to indicate when they can first hear the signal and when they first cannot hear the signal. The program is then switched to a “slow” detailed mode where it does a slower and finer tuned sweep around each of the two start/stop frequencies.
Design a hearing device that will shift or compress the audio range a normal human can hear in order to allow the user to hear frequencies that they no longer hear due to sensorineural hearing loss. Our device will find the minimum and maximum frequencies each individual can hear and will rearrange the normal bandwidth to fit within this range.
For input, we will use an electrot condenser microphone for its tiny size. Sound signal is received by the microphone will be connected to the C5509a DSP chip, which has a built-in analog to digital converter. Then the digital signal is to be modified, in ways of compressing or shifting the frequency, so that all the speech frequency will be in hearing range of the person with sensorineural hearing loss.After the modification, the processed digital signal is converted to an analog signal using the TLV320DAC32 Low-Power Stereo DAC, which will send the signal to an output, TPA6100A2 Headphone Audio Amplifier
The authors would like to thank advisor Dr. Thomas Yen for his continual help throughout the semester.