Department of Phonetics and Linguistics

SiVo -3: A NEW TYPE OF SPEECH-PATTERN HEARING AID FOR PROFOUNDLY HEARING IMPAIRED PEOPLE

Jianing WEI & Kerensa SMITH


1. Introduction
A speech-pattern hearing aid for profoundly hearing impaired people has been developed for some years in this Department. The first generation of the SiVo aid (Sinusoidal Voice) extracted the voice fundamental frequency from speech and presented it as a sinusoid with constant amplitude. The second generation of the SiVo device provided not only the voice fundamental frequency but also the speech amplitude envelope information. The output sinusoid followed the voice fundamental frequency and was amplitude-modulated by the speech amplitude envelope. Now a further speech pattern element, voiceless frication information, is being incorporated in the latest generation of the device SiVo-3. This speech-analytic approach has proved to be an effective auditory supplement to lipreading for profoundly hearing impaired listeners (Rosen et al., 1979, Faulkner et al., 1992, Wei 1993a). As well as this further development, SiVo-3 also additionally functions as a conventional amplifying hearing aid which basically amplifies the input signal. The patient can choose between the analytic mode and the conventional amplifying mode of the device.

This paper will cover four areas:

1. The function of the SiVo-3 device.

2. Fitting of SiVo-3.

3. Signal processing strategies of SiVo-3.

4. The field trials of SiVo-3 and some preliminary results.

2. Function of the Device
SiVo-3 is a digital processing hearing aid. The real-time software can be programmed onto the device via its interface with a PC. The function of the device depends on the real-time code stored on it. The current real-time software for the SiVo-3 device has combined both the speech-analytic hearing aid function and the conventional amplifying hearing aid function. They are chosen by the positions of two front-panel switches "switch between modes" and "multi-purpose switch" of the device as shown in Figure 1:


Figure 1: The SiVo-3 Hearing Aid.

Table 1: The functions of the two front panel switches.

Switch
between
Modes
(Left
Switch)
Multipurpose
Switch
(Right
Switch)
Function
up/middle
up/middle
up/middle
down
down
up
middle
down
up/down
middle
normal SiVo 3 operation
SiVo 3, mapitch (pitch shifted down by) 50 Hz
SiVo 3, mapitch 80 Hz
conventional hearing aid, normal operation
conventional hearing aid, 1kHz 18dB/octave highpass filtered

2.1 Speech-Analytic Mode
In the speech-analytic mode, the device provides speech fundamental frequency, amplitude and voiceless frication information extracted from speech. The voice fundamental frequency information is presented as a sinusoid following the voice pitch. The voiceless frication is detected when there is a voiceless fricative/affricate sound such as and . It also detects the strong plosive sounds and This information is presented as a low frequency noise within 2000 Hz. The spectrum of the noise is shaped by a filter matching the patient's most comfortable level (MCL) between 125 and 2000 Hz. The speech amplitude envelope is used to amplitude-modulate both the sinusoid and the low frequency noise. The output level of the sound is guaranteed to be within the patient's dynamic range. It is possible to switch off the fundamental frequency or frication sound separately by changing the position of a rotary switch on the processor circuit board. The rotary switch has 16 positions marked 0 - F.


Rotary Switch Position
Function
0
normal SiVo-3 operation
2
frication sound output only
4
SiVo 2 mode which generates sinusoid only,
frication sound switched off.

The voice fundamental frequency can be shifted down by 50 Hz or 80 Hz if needed. This is useful for patients who have problems hearing a female or child's voice. This function is determined by the position of the multipurpose switch, as summarised in Table 1.

2.2 Conventional Amplifying Mode
In the conventional amplifying mode, the hearing aid acts like an amplifier. The specification for this mode was provided by Oticon A/S. The maximum output level can reach 148 dB SPL. A highpass filter with 1 kHz/octave low frequency cut designed to reduce low frequency noise can be switched on and off by changing the position of the multi-purpose switch as shown in Table 1. The insertion gain for each frequency is set from thresholds measured by the setting-up software according to the POGO fitting rule (McCandless & Lyregaard, 1983). A filter that matches the patient's insertion gain across the 2000 Hz frequency band is applied to the input signal. An earphone compensation filter is also applied to flatten the frequency response peaks caused by resonance in the earphone.

3. Fitting the SiVo-3 Device
One of the important features of the SiVo-3 device is that it can store the patient's audiogram data (the threshold level, comfortable level, and the discomfort level across different frequencies) and the matching filter's parameters via its set-up software SiVoSet1 and the device's PC interface. Therefore the SiVo-3 device can be tailored for each individual patient. The real-time code guarantees that the output signal of the hearing aid matches the specified data of the patient.

1The SiVoSet software was originally developed by Julian Daley and John Walliker, Dept. of Phonetics and Linguistics, University College London.
The SiVoSet software allows the user to measure the patient's pure tone audiogram. The pure tone sound is played out through the device. Three levels (threshold, comfortable, and discomfort levels) are measured at different frequencies. A matching filter is then selected by the SiVoSet software to match the comfortable curve of the patient. The SiVoSet can also generate a noise which is spectrum-shaped by the matching filter. Therefore the threshold, comfortable, and discomfort level for the noise can be measured similar to the pure tone measurement. The matching filter's parameters together with the patient's pure tone audiogram data and the three levels for the noise sound are all stored on SiVo-3.

In the speech-analytic mode, the output level of the sinusoid is limited within the specified dynamic range of the patient at each measured frequency specified by SiVoSet. The level of the low frequency noise output representing voiceless frication is also limited by the measured threshold and discomfort level for the same noise.

In the conventional amplifying mode, a filter is selected to provide the frequency-gain characteristic required by the POGO fitting rule. The filter parameters are loaded onto the SiVo-3 device. The average discomfort hearing level is also specified and stored onto the device to limit the maximum output level.

4. Speech Signal Processing Strategies for the SiVo-3 Hearing Aid

4.1 The Conventional Amplifying Mode
The speech signal processing task in the conventional amplifying mode is relatively simple compared with the speech pattern extraction algorithms employed in the speech-analytic mode of the SiVo-3 device.

The sampling rate of the device in the conventional mode is 4 kHz. The information above 2 kHz is cut off. An LED light on the front panel will be flash if the level of the input signal to the A/D converter is too high. When the multi-purpose switch is at the middle position, a 1 kHz 18dB/octave high-pass filter is applied to the sampled signal. An FIR matching filter that matches the required POGO gains with its parameters specified by the SiVoSet software is applied. After filtering, the positive and negative peaks are limited to ensure the output signal doesn't exceed the averaged discomfort hearing level. The final stage is to apply an earphone compensation filter to flatten the frequency response peaks caused by resonance in the earphone.

4.2 Speech-analytic Pattern Extraction
In the speech-analytic mode of the SiVo-3, the device's sampling rate is 16 kHz. The voice fundamental frequency/period and voiceless frication information are extracted from speech. The performance of these pattern extractors directly affects the overall effectiveness of the hearing aid. The most important issue is then how to develop robust and fast algorithms for extracting these speech patterns.

4.2.1 Real-Time Voice Fundamental Frequency Extraction Algorithm
The voice fundamental frequency extractor consists of three parts:

1) a pre-processor.

2) a main extractor.

3) a post-processor.

The pre-processor is a 4th order low-pass filter with cut-off frequency at 800 Hz. The input signal to the pre-processor is sampled at 8 kHz. The output from the pre-processor is down-sampled at 2 kHz before it feeds into the main fundamental frequency extractor.

The main fundamental frequency extractor is a multi-layer perceptron pattern classifier (Rumelhart & McClelland, 1988). Pattern classification is the process by which the input vectors are classified into significantly different categories. The system that performs this function is called a pattern classifier. Neural network models are specified by three elements: 1) Net topology; 2) Node characteristics; and 3) Training and learning rules. The nonlinearity used within nodes is one important factor in the capabilities of the neural network models. In supervised training, the neural nets are provided with information or labelling that specifies the correct class for new input patterns during training. Multi-layer perceptrons (MLP) are "feed forward" nets with one or more hidden layers between the input and output nodes.

The MLP fundamental period extractor for the SiVo-3 device as shown in Figure 2 consists of four layers: an input layer, two hidden layers and an output layer. The input layer has 61 nodes, the first hidden layer 20 nodes, the second hidden layer 6 nodes, and the output layer one node. Each layer is fully-connected with the adjacent layer(s).



Figure 2: Real-Time MLP Fundamental Period Extractor.

The training data for the MLP algorithm were recorded in a non-acoustically treated room at UCL, with a 50 cm speaker to microphone spacing. They contain two signal channels (speech and Laryngograph). Eight speakers (four male and four female) were asked to read two passages "Arthur the Rat" and "The Rainbow Passage". A speech spectrum shaped noise was added to part of the training data with various SNR (speech max rms over 500 ms window / noise rms) at 20 dB, 15 dB, and 10 dB. Then the mixed training data were attenuated by 0 dB, 6 dB, and 12 dB, in order to present different signal levels to the MLP classifier so that the classifier would be less sensitive to the input signal level.

The target data is a normalised and half-wave rectified Laryngograph signal (Lx).

The MLP training is accomplished by sequentially applying input vectors, while adjusting network weights according to a predetermined learning procedure. During training, the network weights gradually converge to values such that each input vector produces the desired output vector. Supervised training requires the pairing of each input vector with a target vector representing the desired target output. The success of the training depends not only on the structure of the MLP, but also on the accuracy of the target values used in the training. Figure 3 displays a sample of speech waveform, the input and output of the MLP extractor and the target data.


Figure 3:
(1) Speech waveform.
(2) MLP target data: normalised and half-wave rectified Lx signal.
(3) MLP input data.
(4) MLP output data.
(5) output from the post-processor.

After the MLP model was trained on the UNIX machine, a set of weights were obtained which were scaled in order to be loaded onto the SiVo 3 hardware.

The post-processing stage converts the output from the basic extractor into a desirable format for generating the sinusoid output on the SiVo-3 and also performs error correction and smoothing of the raw estimates from the main extractor.

In the SiVo-2 device, the sinusoid generator repeated the previous sinusoid cycle if it had not received a new fundamental period value at the time when the previous sinusoid cycle has finished. This problem has now been overcome on the SiVo-3 by introducing a buffer in the post-processor, which is 20.5 ms in length (i.e. it holds 41 MLP output values), as shown in Figure 4:

.


Figure 4: Buffer Post-Processor.

The new MLP output will be stored at the beginning of the buffer and all the other values in the buffer will be shifted along the buffer. The value at the end of the buffer will be shifted out each 0.5 ms when a new MLP value enters. When the value at the end of buffer is a peak above the output threshold, the processor will look along the buffer for a second peak. If there is a second peak within the buffer, the time difference between the first and second peaks is calculated. If this fundamental period Tx is the first period after a silent or a voiceless speech segment, the value is sent to the sinusoid generator. If it is during a voicing speech segment, it is then compared with the estimated previous fundamental period Te (the average of the previous two fundamental periods: (Tx-1 + Tx-2)/2 ). If Tx is within the (1± 20%) Te range, then the Tx is sent to the sinusoid generator. If Tx is outside the (1± 20%) Te range, then the processor will look for a different second peak within the (1± 20%) Te range with a lower output threshold. If a peak has been found within this range, then it is used to calculate the present Tx. Otherwise, the previously detected second peak is used to calculate the Tx value.

The MLP fundamental frequency extractor and the buffer processor introduce 35 ms delay in total between the input and output signals of the SiVo. However, studies using normal listeners have shown that if the time delay of the acoustical signal is within 40 ms, it will not affect performance in aided lipreading. (McGrath & Summerfield, 1985).

The MLP fundamental period extractor was compared with other algorithms such as peak-picker(Howard & Fourcin, 1983), Gold-Rabiner method (Gold & Rabiner, 1967), Cepstral analysis (Noll, 1967), the SIFT algorithm and Sub-harmonic Summation method (Hermes, 1988). It performed better in noisy conditions than the other methods (Wei et al, 1993b).

4.2.2 Real-Time Voiceless Frication Detector
The SiVo 3 hardware samples the input speech at a much wider sampling frequency range than the SiVo 2 device. This enables the real-time implementation of the frication detection algorithm since they were initially developed on the SUN SPARC station at UCL using 16 kHz sampling rate. The real-time frication detection algorithm currently implemented on SiVo 3 is a spectral-balance algorithm shown in Figure 5.


Figure 5: Spectral-Balance Frication Detection Algorithm.

When the frication information is detected using the spectral-balance algorithm, it is important to encode the information in an appropriate way in order to match individual patient's residual hearing ability. At present, the noise is software-generated using an on-line noise generator. The spectrum of the noise is determined by the patient's matching filter specified by the SiVoSet software. The level of the noise is also determined by the SiVoSet measurement of the patient's dynamic range for this noise.

If the voice fundamental period extractor has detected a fundamental period, the output of the frication detector is ignored.

5. Field trials of the SiVo-3 device
The SiVo-3 device is currently being tested in field trials as part of the OSCAR project: an EC funded TIDE (Technical Initiatives for the Disabled and Elderly) project. This involves partners in France, the Netherlands, Sweden, and the UK and field trials are and will be taking place in each of these four countries. The aims of the field trials are as follows:

1. Investigate the value of a dual-mode aid which is designed for both general use and face-to-face communication.

2. Examine the added benefit of speech analytic processing over and above conventional amplification.

3. Compare unimodal and bimodal (i.e. both auditory and tactile) stimulation.

4. To investigate how well the speech analytic processing method performs in noise during audio visual consonant identification tests and sentence identification tests. Previous research in the STRIDE project (Faulkner, 1994, 1996) gave good results for SiVo II in noise in similar tests. It is hoped that SiVo-3 will give better results because of the post-processing buffer now present.

There will be two phases of the field trials. Aims 1 and 2 will be looked at during the first phase and this is currently underway. The third aim will be investigated during the second phase and the fourth aim throughout the trials.

The basic outline of the first phase of the field trials is shown below.

Table 2: Outline of the first phase of the field trials
1 month3 months1 month
Pretraining assessmentsTraining in both the conventional and analytic modes and home use of the aid - a minimum of 6 sessions Posttraining assessments

5.1 Pretraining Assessments
Both speech perception and speech production are assessed. During the pretraining assessment, each subject is assessed in 4 conditions:
a) No hearing aid.
b) Own hearing aid.
c) Analytic mode of SiVo-3.
d) Conventional mode of SiVo-3.

The speech perception tests are as follows:

  1. Consonant identification test (audio-visual). 14 consonants are presented randomly 4 times within one list in the usual vCv format. (The UK presented the consonants 3 times in each list). The consonants are common to all four languages involved; i.e. . The lists are presented twice in each condition.
  2. Stress placement tests (audio alone). The subject has to locate the nuclear stress in a simple 3-word sentence. Two sentences are used: a completely voiced sentence which, in the UK, is "We all know" and a sentence containing voiceless consonants e.g."She should try". One list consists of 7 random repetitions of each version of the sentence (i.e. 21 tokens per list). Two lists are presented per condition apart from the unaided condition.
  3. Sentence identification test. (audio-visual). In the UK this is the BKB sentence identification test. Again, two lists are presented per condition.

Speech production changes are also measured. Recordings are made during the pretraining and posttraining assessments and also half way through the training period. The subject is recorded producing a sustained vowel on a level tone, reading a short passage and reading a word list.
5.2 Selection Criteria for Subjects
Subjects are selected according to the following criteria:

  1. Subjects are postlingually deafened adults whose loss is sensori-neural in origin.
  2. The better ear should have hearing levels as follows:
  3. a) The pure tone average across 250, 500 and 1k Hz should be greater than 90dB.
  4. b) Where there is measurable hearing at 2kHz the pure tone average across 500, 1k and 2k should be at least 100dB.
  5. c) The threshold of hearing at 2kHz should be over 100dB.
  6. d) There should be no hearing at 4kHz.
  7. The subject should have no prior experience of a speech analytic processing aid like the SiVo.
  8. The age range is 18 to 75 years.
  9. Those subjects who have a partner or friend who is able to spend time training the subject at home with materials supplied by the therapists involved are given priority.
  10. The subjects should be well motivated to take part in a research project.
  11. They should be able to manage the controls of the SiVo aid.

6. Preliminary results
At the time of writing, only results from the pretraining assessments carried out in the UK are available. These are from four subjects selected according to the above criteria. Table 3 shows these results.

More detailed results from the vCv identification test using SINFA analysis are shown in Table 4. No analysis of frication has been done yet.

The speech production results are as yet unanalysed.

As each subject's results are so different, they will be considered separately.

Table 3: Preliminary results from UK subjects - pretraining speech perception tests
PatientHearing
device
VCVs %
overall corr.
She should try. % corr. We all know. % corr.BKB sentences %
BHAnal. mode70.25 97.571.4T=64 L=66
Conv. mode65.5 50.035.7T=68 L=71
Own aid58.5 57.135.7T=43 L=49
No aid37.0 __________________ T=36 L=39
IKAnal. mode44.0 19.042.9T=19 L=32
Conv. mode52.4 33.340.5T=63 L=76
Own aid57.2 71.450.0T=63 L=80
No aid33.4 __________________ T=41 L=49
PRAnal. mode35.7 38.145.3T=6 L=10
Conv. mode26.2 26.233.3T=18 L=19
Own aid_________ __________________ _________
No aid38.1 __________________ T=4 L=5
BWAnal. mode39.3 21.428.6T=37 L=41
Conv. mode50.0 31.019.0T=48 L=52
Own aid_________ __________________ _________
No aid46.5 __________________ T=54 L=58

  • T = Tight scoring L = Loose scoring (Bamford and Wilson, 1979)
  • Anal. mode = Analytic mode of SiVo-3
  • Conv. mode = Conventional mode of SiVo-3
  • Table 4: More detailed analyses of the vCv results
    PatientHearing device Voicing
    % info. transfer
    Place
    % info transfer
    Manner
    % info transfer
    BHAnal. mode76.3 95.658.8
    Conv. mode57.2 90.043.5
    Own aid24.2 87.057.0
    No aid9.7 85.541.1
    IKAnal. mode23.2 79.032.9
    Conv. mode34.4 79.639.9
    Own aid29.9 88.552.1
    No aid4.9 69.830.3
    PRAnal. mode12.4 59.127.3
    Conv. mode3.8 52.121.9
    Own aid_________ _____________________
    No aid19.7 69.028.4
    BWAnal. mode20.1 71.142.1
    Conv. mode16.9 71.946.6
    Own aid__________ ____________________
    No aid14.9 75.045.9

    BH: This subject found the information conveyed by the analytic mode most beneficial in both the vCv identification tests and the stress placement tests. Closer examination of the vCv results (Table 4) shows more clearly that the benefits are particularly in perception of manner and most strikingly voicing information. In comparison with his own aid the voicing % information transfer scores are approximately three times better with the analytic (as opposed to just twice as good with the conventional mode). He is a good lipreader and so , although he shows an improvement in place % information transfer with the analytic mode, this is quite small because his scores are all high in this area.

    The extracted elements of voice fundamental frequency and amplitude available in the analytic mode of the SiVo seem to greatly enhance BH's ability to correctly detect nuclear stress. In fact, his scores are almost twice as good in this mode compared to either the conventional mode of the SiVo or his own aid. It was apparent during the testing session that he was more relaxed when using the analytic mode.

    There is little difference, in the BKB results, between those obtained with the analytic mode of the SiVo and those with the conventional mode. However, these are considerably better than the two other conditions suggesting that both modes give more information than his own aid.

    IK: This subject gained better results on the vCv tests and stress placement tests with his own hearing aid. This is not uncommon at the pretraining testing stage despite subjects' apparent dislike of their own aid(s). Often, subjects have many years of experience with a particular device and have learnt to use it to the best of their ability even if this is not satisfactory. Therefore, when they switch to a new aid the sound is unfamiliar and the subject has to learn to use the new information presented to them. In IK's case he has been wearing his current aid in his left ear for 2 years.

    Comparison of the two modes of the SiVo reveals interesting results. When IK was first fitted with the SiVo he reported that he did not like the analytic mode at all. This subjective judgement is very much reflected in his results. His scores are considerably higher with the conventional mode, particularly in the voiceless stress placement tests and the BKB sentence tests.

    PR: This subject has no hearing aid of her own and has not worn aids for 30 years. Table 3 shows that at present she is not able to make use of the extra information presented in either mode of the SiVo to help her in the identification of consonants. In fact, her vCv results are better when she is using no aid at all. This is true even for the amount of voicing she is able to detect. It could be that following 30 years of no auditory stimulation she finds any sounds she hears confusing. It is felt that with sufficient training she could learn to interpret the new auditory cues.

    This hope is not unfounded as the results for the stress placement tests are approximately 12% higher with the analytic mode than the conventional mode of the SiVo. This suggests that she is making some use of the voice fundamental frequency and amplitude cues delivered through the analytic mode, even though it shows a small effect at present.

    Her BKB results are best with the conventional mode of the SiVo. However, these are still very low and so it is not possible to draw any conclusions from these results.

    BW: This subject has not worn hearing aids for 7 years. Table 3 shows that BW is best able to identify consonants with the conventional mode. A breakdown of the vCv results in Table 4, however, shows that she gains slightly more voicing information with the analytic mode. This is to be expected with the cues presented.

    Her stress placement results are low in both modes and are inconclusive. She found this particular set of tests very difficult.

    Her BKB results show a slight improvement from the analytic mode to the conventional mode and then to unaided. As with subject PR it is possible that the addition of auditory cues has actually detracted from BW's ability to lipread. Again, it is felt that with training these extra cues will enhance BW's ability to lipread.

    7. Conclusions
    At this stage it is very difficult to show conclusive results. In fact, the purpose of the pretraining tests is to provide a basis for comparison with later tests once a program of training has been completed. However, it is possible to see a few interesting trends emerging. One such trend is that 3 out of the 4 subjects gain more voicing information with the analytic mode of the SiVo than the conventional mode albeit still a small amount in three of them. This increase is to be expected with the presentation of fundamental frequency as one of the extracted elements. Another trend shown is that all four subjects perform better on the BKB sentence tests with the conventional mode than the analytic mode. In fact, in three subjects' results, the conventional mode yields the best results against all the other conditions. This could be because the sound is more natural with the conventional mode.

    8. Future Work
    Following these tests, each subject will now partake in a minimum of six sessions of training. During this time the subjects will be trained to make the best use of the information available to them with the analytic and conventional modes of the SiVo. Training covers consonant identification, stress placement, sentence work and Connected Discourse Tracking (CDT). The subjects will be asked to keep a daily diary in which they will record the situations they use each mode in and what sounds they can hear. From these it will be possible to assess the benefit for the subjects of having a dual-mode aid. When the training is completed, the subjects will undergo more testing. This will consist of a more extended period of the same type of tests carried out as above.

    The field trials are in the early stages and still much work has to be done. However, these preliminary results are promising and suggest an encouraging outcome.

    Acknowledgement
    This work is supported by the European Commission's OSCAR (Optimal Speech Communication Assistance for Residual Abilities) project within the TIDE (Technology for Integration of the Disabled and Elderly) program.

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    Wei J , Howells D, Fourcin A, and Faulkner A (1993b) Larynx period and frication detection methods in speech pattern hearing aids, Speech Hearing and Language, Work in Progress, Dept. of Phonetics and Linguistics, University College London, 7, 269-276.

    © 1996 Jianing Wei and Kerensa Smith


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