scholarly journals Effect of the Target and Conflicting Frequency and Time Ranges on Consonant Enhancement in Normal-Hearing Listeners

2021 ◽  
Vol 12 ◽  
Author(s):  
Yang-Soo Yoon
Keyword(s):  
Signal Processing ◽  
Normal Hearing ◽  
Peak Performance ◽  
Time Range ◽  
Frequency Range ◽  
Target Condition ◽  
Target Frequency ◽  
Target Time ◽  
Consonant Recognition ◽  
Low Pass

In this paper, the effects of intensifying useful frequency and time regions (target frequency and time ranges) and the removal of detrimental frequency and time regions (conflicting frequency and time ranges) for consonant enhancement were determined. Thirteen normal-hearing (NH) listeners participated in two experiments. In the first experiment, the target and conflicting frequency and time ranges for each consonant were identified under a quiet, dichotic listening condition by analyzing consonant confusion matrices. The target frequency range was defined as the frequency range that provided the highest performance and was decreased 40% from the peak performance from both high-pass filtering (HPF) and low-pass filtering (LPF) schemes. The conflicting frequency range was defined as the frequency range that yielded the peak errors of the most confused consonants and was 20% less than the peak error from both filtering schemes. The target time range was defined as a consonant segment that provided the highest performance and was decreased 40% from that peak performance when the duration of the consonant was systematically truncated from the onset. The conflicting time ranges were defined on the coincided target time range because, if they temporarily coincide, the conflicting frequency ranges would be the most detrimental factor affecting the target frequency ranges. In the second experiment, consonant recognition was binaurally measured in noise under three signal processing conditions: unprocessed, intensified target ranges by a 6-dB gain (target), and combined intensified target and removed conflicting ranges (target-conflicting). The results showed that consonant recognition improved significantly with the target condition but greatly deteriorated with a target-conflicting condition. The target condition helped transmit voicing and manner cues while the target-conflicting condition limited the transmission of these cues. Confusion analyses showed that the effect of the signal processing on consonant improvement was consonant-specific: the unprocessed condition was the best for /da, pa, ma, sa/; the target condition was the best for /ga, fa, va, za, ʒa/; and the target-conflicting condition was the best for /na, ʃa/. Perception of /ba, ta, ka/ was independent of the signal processing. The results suggest that enhancing the target ranges is an efficient way to improve consonant recognition while the removal of conflicting ranges negatively impacts consonant recognition.

Intelligibility of Filtered Synthetic Sentences

1967 ◽  
Vol 10 (2) ◽  
pp. 289-298 ◽  
Author(s):  
Charles Speaks
Keyword(s):  
Low Frequency ◽  
Normal Hearing ◽  
Frequency Filtering ◽  
Frequency Range ◽  
Frequency Bands ◽  
Low Pass ◽  
High Pass

The effects of frequency filtering on intelligibility of synthetic sentences were studied on three normal-hearing listeners. Performance-intensity (P-I) functions were defined for several low-pass and high-pass frequency bands. The data were analyzed to determine the interactions of signal level and frequency range on performance. Intelligibility of synthetic sentences was found to be quite dependent upon low-frequency energy. The important frequency for identification of the materials was approximately 725 Hz. These results are compared with previous findings concerning the intelligibility of single words in quiet and in noise.

scholarly journals Enhancement of Consonant Recognition in Bimodal and Normal Hearing Listeners

2019 ◽  
Vol 128 (6_suppl) ◽  
pp. 139S-145S
Author(s):  
Yang-Soo Yoon ◽  
Britteny Riley ◽  
Henna Patel ◽  
Amanda Frost ◽  
Paul Fillmore ◽  
...  
Keyword(s):  
Signal Processing ◽  
Normal Hearing ◽  
Auditory Training ◽  
Listening Condition ◽  
Training Tool ◽  
3 Dimensional ◽  
Place Of Articulation ◽  
Consonant Recognition ◽  
Before And After ◽  
Binaural Listening

Objectives: The present study investigated the effects of 3-dimensional deep search (3DDS) signal processing on the enhancement of consonant perception in bimodal and normal hearing listeners. Methods: Using an articulation-index gram and 3DDS signal processing, consonant segments that greatly affected performance were identified and intensified with a 6-dB gain. Then consonant recognition was measured unilaterally and bilaterally before and after 3DDS processing both in quiet and noise. Results: The 3DDS signal processing provided a benefit to both groups, with greater benefit occurring in noise than quiet. The benefit rendered by 3DDS was the greatest in binaural listening condition. Ability to integrate acoustic features across ears was also enhanced with 3DDS processing. In listeners with normal hearing, manner and place of articulation were improved in binaural listening condition. In bimodal listeners, voicing and manner and place of articulation were also improved in bimodal and hearing aid ear–alone conditions. Conclusions: Consonant recognition was improved with 3DDS in both groups. This observed benefit suggests 3DDS can be used as an auditory training tool for improved integration and for bimodal users who receive little or no benefit from their current bimodal hearing.

Second Formant Transitions of Selected Consonant-Vowel Combinations in the Speech of Deaf and Normal-Hearing Children

1976 ◽  
Vol 19 (2) ◽  
pp. 279-289 ◽  
Author(s):  
Randall B. Monsen
Keyword(s):  
Speech Intelligibility ◽  
Normal Subjects ◽  
Normal Hearing ◽  
Hearing Impaired ◽  
Acoustic Cues ◽  
Target Frequency ◽  
Second Formant ◽  
Hearing Children

Although it is well known that the speech produced by the deaf is generally of low intelligibility, the sources of this low speech intelligibility have generally been ascribed either to aberrant articulation of phonemes or inappropriate prosody. This study was designed to determine to what extent a nonsegmental aspect of speech, formant transitions, may differ in the speech of the deaf and of the normal hearing. The initial second formant transitions of the vowels /i/ and /u/ after labial and alveolar consonants (/b, d, f/) were compared in the speech of six normal-hearing and six hearing-impaired adolescents. In the speech of the hearing-impaired subjects, the second formant transitions may be reduced both in time and in frequency. At its onset, the second formant may be nearer to its eventual target frequency than in the speech of the normal subjects. Since formant transitions are important acoustic cues for the adjacent consonants, reduced F 2 transitions may be an important factor in the low intelligibility of the speech of the deaf.

Measurement of Cr(VI) Based on Absorption Spectrum through Plastic Fiber

2014 ◽  
Vol 687-691 ◽  
pp. 948-951
Author(s):  
Wei Jun Hu
Keyword(s):  
Signal Processing ◽  
Absorption Spectrum ◽  
Optic Fiber ◽  
Signal Process ◽  
Linear Fitting ◽  
Non Linear ◽  
Low Pass ◽  
Measurement Experiment

Considering the advantage of optic fiber, a methods of measuring Cr (VI) based on absorption spectrum through plastic fiber is introduced, which includes structure of measurement, experiment process, spectrum signal process. After signal processing based on low-pass filtering and non-linear fitting, five concentrations of Cr (VI) can be differed easily and the peak values of spectrum corresponding to five concentrations accord with the Longbow Bill's law . In this way, the measurement concentration can limit down to 0. 0.0660 μg/ml.

scholarly journals Eliminating unwanted signals in sound by using digital signal processing system

2020 ◽  
Vol 18 (2) ◽  
pp. 829
Author(s):  
Amer T Saeed ◽  
Zaid Raad Saber ◽  
Ahmed M. Sana ◽  
Musa A. Hameed
Keyword(s):  
Signal Processing ◽  
Digital Signal Processing ◽  
Digital Filters ◽  
Processing System ◽  
Digital Signal ◽  
Noise Signal ◽  
Time Range ◽  
Signal Processing System ◽  
Digital Signal Processing System ◽  
Sound Files

<p><a name="_Hlk536186602"></a><span style="font-size: 9pt; font-family: 'Times New Roman', serif;">Unwanted signals or noise signals in sound files are considered one of the major challenges and issues for a thousand users. It is impossible to reduce or remove these noise signals without identifying their types and ranges. Therefore, to address one of the big problems in the digital or analogue communication, which is noise signals or unwanted signals, an adaptive selection method and noise signal removal algorithm are proposed in this research. The proposed algorithm is done through specifying the types of undesirable signals, frequency, and time range, then utilizing digital signal processing system which includes design several types of digital filters based on the types and numbers of unwanted signals. Four digital filters are used in this research to remove noise signals from the sound file by implementing the proposed algorithm using Matlab Code. Results show that our proposed algorithm was done successfully and the whole noise signals were removed without any negative consequence in the output sound signal. </span><span style="font-family: 'Times New Roman', serif; font-size: 9pt;">Unwanted signals or noise signals in sound files are considered one of the major challenges and issues for a thousand users. It is impossible to reduce or remove these noise signals without identifying their types and ranges. Therefore, to address one of the big problems in the digital or analogue communication, which is noise signals or unwanted signals, an adaptive selection method and noise signal removal algorithm are proposed in this research. The proposed algorithm is done through specifying the types of undesirable signals, frequency, and time range, then utilizing digital signal processing system which includes design several types of digital filters based on the types and numbers of unwanted signals. Four digital filters are used in this research to remove noise signals from the sound file by implementing the proposed algorithm using Matlab Code. Results show that our proposed algorithm was done successfully and the whole noise signals were removed without any negative consequence in the output sound signal.</span></p>

scholarly journals Design of Reconfigurable Time-to-Digital Converter Based on Cascaded Time Interpolators for Electrical Impedance Spectroscopy

Sensors ◽  
2020 ◽  
Vol 20 (7) ◽  
pp. 1889
Author(s):  
Sounghun Shin ◽  
Yoontae Jung ◽  
Soon-Jae Kweon ◽  
Eunseok Lee ◽  
Jeong-Ho Park ◽  
...  
Keyword(s):  
Impedance Spectroscopy ◽  
High Frequency ◽  
Electrical Impedance ◽  
Phase Error ◽  
Electrical Impedance Spectroscopy ◽  
Time Range ◽  
Digital Converter ◽  
Frequency Range ◽  
Input Time ◽  
Interpolation Factor

This paper presents a reconfigurable time-to-digital converter (TDC) used to quantize the phase of the impedance in electrical impedance spectroscopy (EIS). The TDC in the EIS system must handle a wide input-time range for analysis in the low-frequency range and have a high resolution for analysis in the high-frequency range. The proposed TDC adopts a coarse counter to support a wide input-time range and cascaded time interpolators to improve the time resolution in the high-frequency analysis without increasing the counting clock speed. When the same large interpolation factor is adopted, the cascaded time interpolators have shorter measurement time and smaller chip area than a single-stage time interpolator. A reconfigurable time interpolation factor is adopted to maintain the phase resolution with reasonable measurement time. The fabricated TDC has a peak-to-peak phase error of less than 0.72° over the input frequency range from 1 kHz to 512 kHz and the phase error of less than 2.70° when the range is extended to 2.048 MHz, which demonstrates a competitive performance when compared with previously reported designs.

scholarly journals Improved Performance of M-Class PMUs Based on a Magnitude Compensation Model for Wide Frequency Deviations

Mathematics ◽  
2020 ◽  
Vol 8 (8) ◽  
pp. 1361
Author(s):  
Jose Roberto Razo-Hernandez ◽  
Ismael Urbina-Salas ◽  
Guillermo Tapia-Tinoco ◽  
Juan Pablo Amezquita-Sanchez ◽  
Martin Valtierra-Rodriguez ◽  
...  
Keyword(s):  
Power Systems ◽  
Frequency Range ◽  
Magnitude Response ◽  
Modulation And Demodulation ◽  
Variable Function ◽  
Low Pass ◽  
Measurement Units ◽  
Phasor Estimation ◽  
Improved Performance ◽  
Reporting Rates

Phasor measurement units (PMUs) are important elements in power systems to monitor and know the real network condition. In order to regulate the performance of PMUs, the IEEE Std. C37.118.1 stablishes two classes—P and M, where the phasor estimation is carried out using a quadrature oscillator and a low-pass (LP) filter for modulation and demodulation, respectively. The LP filter plays the most important role since it determines the accuracy, response time and rejection capability of both harmonics and aliased signals. In this regard and by considering that the M-class filters are used for more accurate measurements, the IEEE Std. presents different M-class filters for different reporting rates (when a result is given). However, they can degrade their performance under frequency deviations if the LP frequency response is not properly considered. In this work, a unified model for magnitude compensation under frequency deviations for all the M-class filters is proposed, providing the necessary values of compensation to improve their performance. The model considers the magnitude response of the M-class filters for different reporting rates, a normalized frequency range based on frequency dilation and a fitted two-variable function. The effectiveness of the proposal is verified using both static and dynamic conditions for frequency deviations. Besides that, a real-time simulator to generate test signals is also used to validate the proposed methodology.

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