This document presents a low-power, high-speed amplification filterbank designed for digital hearing aids. The researchers developed a 10-ms 18-band quasi-ANSI S1.11 1/3-octave filterbank to process 24 kHz audio signals with reduced group delay and computational complexity compared to the standard. Simulation results showed the filterbank could compensate for various hearing loss prescriptions with less than 1.5 dB maximum matching error. The filterbank was also implemented in a 90 nm CMOS process, consuming only 73 uW of power, or 27 uW at a reduced supply voltage of 0.6V.
3. speech processing algorithms for perception improvement of hearing impaire...k srikanth
This document presents a novel algorithm to improve speech perception for hearing impaired patients using dichotic speech processing and presentation techniques. The algorithm splits speech into multiple frequency bands using dyadic filters to achieve bands of constant bandwidth, 1/3 octave bandwidth, and critical bandwidth. Listening tests on 5 subjects with hearing loss were conducted using 15 syllable speech material processed with the different filter sets. The results showed that processing with 1/3 octave bands improved recognition scores and reduced response times the most compared to unprocessed speech and speech processed with the other two filter sets. The algorithm aims to overcome issues like spectral and temporal masking that impair speech perception for those with sensorineural hearing loss.
To Improve Speech Recognition in Noise for Cochlear Implant UsersIOSR Journals
This document describes two experiments that test hypotheses about how to improve speech recognition in noise for cochlear implant users. The first experiment investigates whether providing access to clean obstruent consonants results in masking release for implant users. The results support this hypothesis, showing improved recognition when obstruents were clean, especially at low signal-to-noise ratios. The second experiment tests whether less compression of unvoiced segments makes acoustic landmarks more evident, again improving noise recognition. The results suggest this selective compression approach provides benefits over traditional logarithmic compression. Overall, the findings indicate that better conveying obstruent and vowel/consonant boundary cues has potential to enhance noise robustness for cochlear implant listeners.
Artificially enhancing better-ear glimpsing cues to improve understanding of ...HEARnet _
Artificially enhancing better-ear glimpsing cues to improve understanding of speech in noise for listeners with hearing loss cues to improve understanding of speech in noise for listeners with hearing loss
The Pure Tone Audiometry (PTA) test is used to determine a person's hearing threshold levels using pure tone pulses presented at standardized frequencies from 125-8000Hz. The threshold is the lowest sound level at which a person responds correctly to 50% of tones. Tones are presented one ear at a time through earphones to obtain the threshold for each ear separately. Special equipment including an oscillator, attenuator and earphones are used to generate and present the tones while masking noise may be used in the non-testing ear to prevent crossover. Multiple factors can influence PTA results including the testing environment, equipment calibration and placement, and the test method used.
This document summarizes a research paper on speech enhancement using the signal subspace algorithm. It begins with an abstract describing how noise degrades speech quality and intelligibility in communication systems. It then provides background on speech enhancement objectives and commonly used methods like spectral subtraction and signal subspace. The paper describes the signal subspace algorithm and shows its ability to enhance speech signals by suppressing noise. Experimental results on sine waves with added Gaussian noise demonstrate improved peak signal-to-noise ratios when using the signal subspace method compared to the noisy signals. The conclusion is that the algorithm removes noise to a great extent from noisy speech.
Implementation of a Digital Hearing Aid with User-Settable Frequency Response...IAMCP MENTORING
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings describes among others the Petralex smartphone App as a hearing aid - the product of the IAMCP members - IT4You (www.petralex.pro)
This is my presentation on a Journal Club. It's based on the article: "Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners". You can find all the references in the slide at the end of the article. I review very basic techniques in noise reduction, and how the techniques are implemented in the area of deep neural-network.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
3. speech processing algorithms for perception improvement of hearing impaire...k srikanth
This document presents a novel algorithm to improve speech perception for hearing impaired patients using dichotic speech processing and presentation techniques. The algorithm splits speech into multiple frequency bands using dyadic filters to achieve bands of constant bandwidth, 1/3 octave bandwidth, and critical bandwidth. Listening tests on 5 subjects with hearing loss were conducted using 15 syllable speech material processed with the different filter sets. The results showed that processing with 1/3 octave bands improved recognition scores and reduced response times the most compared to unprocessed speech and speech processed with the other two filter sets. The algorithm aims to overcome issues like spectral and temporal masking that impair speech perception for those with sensorineural hearing loss.
To Improve Speech Recognition in Noise for Cochlear Implant UsersIOSR Journals
This document describes two experiments that test hypotheses about how to improve speech recognition in noise for cochlear implant users. The first experiment investigates whether providing access to clean obstruent consonants results in masking release for implant users. The results support this hypothesis, showing improved recognition when obstruents were clean, especially at low signal-to-noise ratios. The second experiment tests whether less compression of unvoiced segments makes acoustic landmarks more evident, again improving noise recognition. The results suggest this selective compression approach provides benefits over traditional logarithmic compression. Overall, the findings indicate that better conveying obstruent and vowel/consonant boundary cues has potential to enhance noise robustness for cochlear implant listeners.
Artificially enhancing better-ear glimpsing cues to improve understanding of ...HEARnet _
Artificially enhancing better-ear glimpsing cues to improve understanding of speech in noise for listeners with hearing loss cues to improve understanding of speech in noise for listeners with hearing loss
The Pure Tone Audiometry (PTA) test is used to determine a person's hearing threshold levels using pure tone pulses presented at standardized frequencies from 125-8000Hz. The threshold is the lowest sound level at which a person responds correctly to 50% of tones. Tones are presented one ear at a time through earphones to obtain the threshold for each ear separately. Special equipment including an oscillator, attenuator and earphones are used to generate and present the tones while masking noise may be used in the non-testing ear to prevent crossover. Multiple factors can influence PTA results including the testing environment, equipment calibration and placement, and the test method used.
This document summarizes a research paper on speech enhancement using the signal subspace algorithm. It begins with an abstract describing how noise degrades speech quality and intelligibility in communication systems. It then provides background on speech enhancement objectives and commonly used methods like spectral subtraction and signal subspace. The paper describes the signal subspace algorithm and shows its ability to enhance speech signals by suppressing noise. Experimental results on sine waves with added Gaussian noise demonstrate improved peak signal-to-noise ratios when using the signal subspace method compared to the noisy signals. The conclusion is that the algorithm removes noise to a great extent from noisy speech.
Implementation of a Digital Hearing Aid with User-Settable Frequency Response...IAMCP MENTORING
The article by Saketh Sharma, Nitya Tiwari, and Prem C. Pandey in the "Int. Conf. on Intelligent Human Computer Interaction" proceedings describes among others the Petralex smartphone App as a hearing aid - the product of the IAMCP members - IT4You (www.petralex.pro)
This is my presentation on a Journal Club. It's based on the article: "Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners". You can find all the references in the slide at the end of the article. I review very basic techniques in noise reduction, and how the techniques are implemented in the area of deep neural-network.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
This study investigated how acute low-tone sensorineural hearing loss (ALHL) affects the perception of binaural beats (BBs). The researchers measured the frequency ranges where BBs were not perceived in 22 ALHL patients, 11 recovered ALHL patients, 5 high-tone hearing loss patients, and 14 normal controls. They found the frequency ranges were significantly wider in ALHL patients and narrowed with hearing recovery. The ranges did not differ between high-tone loss patients and controls, suggesting low-frequency damage affects BB perception by degrading frequency coding. Endolymphatic hydrops presence did not correlate with frequency range widths.
The document discusses various topics related to sound reinforcement systems including:
- The basic components of a sound wave and their definitions.
- Different types of input and output transducers used in sound systems.
- Key specifications and measurements used to evaluate audio equipment performance such as frequency response, signal level, and impedance.
- Concepts like loudness, equal loudness contours, and how they relate to human hearing.
- Different types of microphones, their characteristics, and common applications.
- Components and functions of a mixing console like preamps, auxiliary sends, and monitor mixes.
This document discusses various issues that may arise with cochlear implants and potential ways to address patient complaints through programming adjustments. It outlines common complaints such as sound quality issues, discomfort, loudness problems, and facial nerve stimulation. For each complaint, it lists potential programming parameters that could be modified, such as stimulus levels, filter settings, electrode configurations, and coding strategies. It emphasizes considering an acclimatization period and reassessing programming over time before making changes. It also discusses soft failures where performance declines despite normal integrity testing.
This document summarizes and compares different speech enhancement methods for noisy Punjabi speech at the phoneme level. It describes spectral subtraction, Wiener filtering, Kalman filtering, and RASTA methods. It also proposes a hybrid method using features from Wiener and Kalman filtering. Phonemes of Punjabi language and the methodology are explained. Results applying various noise types at different signal-to-noise ratios on phonemes show the proposed method produces the best enhancement compared to other methods.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
This document summarizes information about noise, sound level measurements, and the effects of noise on humans. It discusses how the decibel scale is used to measure sound intensity, how the loudness of sounds is quantified using phons and sones, and how long-term exposure to noise can cause temporary or permanent hearing loss. It also outlines Occupational Safety and Health Administration standards for permissible noise exposure limits in the workplace.
The document describes a study that measured the frequency response and cutoff frequencies of six cell phones for recording/playback and transmission. For recording/playback, the Apple smartphones had the highest cutoff frequencies around 15,000 Hz, followed by Android smartphones around 6,000-11,000 Hz, and conventional phones around 3,000-3,600 Hz. However, for transmission all phones had cutoff frequencies within 6% of each other near 3,400 Hz, the typical range for phone networks. The document provides background on human hearing, speech, telephone systems, frequency response analysis, and defines cutoff frequencies.
This document discusses voice modification techniques. It compares four models for voice modification: LPC, H/S, TD-PSOLA, and MBR-PSOLA. TD-PSOLA relies on pitch-synchronous overlap-add and modifies the speech signal in the time domain based on analysis and synthesis markings. The document implements a voice modification system using TD-PSOLA that can modify vocal tract, pitch, and time scale parameters to change voice quality, such as making a female voice sound more husky or nasal. Results show the algorithm can effectively modify input voices as desired.
Speech processing strategies for cochlear prostheses the past, present and fu...iaemedu
This document provides a review of speech processing strategies used in cochlear implants. It discusses early single-channel implants and limitations. It then describes the development of multi-channel implants and different types of speech processing strategies used, including waveform strategies like Compressed Analog and Continuous Interleaved Sampling, feature extraction strategies that code formant frequencies, and hybrid strategies. Finally, it summarizes commonly used strategies in current commercial cochlear implants and concludes that strategies providing spectral information allow better speech understanding than those extracting speech features alone.
1) The document proposes a single-port wireless transceiver circuit to enable affordable audio-visual communication for small enterprises, addressing the expensive costs of traditional videoconferencing.
2) A key challenge is reducing audio feedback known as the "Larsen effect" that occurs when a microphone picks up sound from its own speaker, causing an annoying acoustic noise.
3) The paper describes an experimental circuit built by the authors that uses a feedback control unit to monitor microphone and speaker levels and suppress the Larsen effect, allowing for clear audio-visual transmission over a single wireless port. Testing showed the acoustic echo was eliminated when this anti-Larsen effect circuit was employed.
The document discusses the evolution of the NAL fitting formula from a linear to a compression formula with three gain targets. The research showed that for gently sloping losses, similar gain was needed at 65dB inputs. NAL aims to equalize loudness across frequencies rather than normalize to unaided levels. This preserves speech spectrum loudness relationships and is theorized to maximize speech understanding. NAL gain targets are calculated based on phonemic information from the Articulation Index. The two key features of NAL are equalizing loudness across frequencies and providing less gain where loss is worst.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Computational Engineering Research(IJCER)ijceronline
International Journal of Computational Engineering Research(IJCER) is an intentional online Journal in English monthly publishing journal. This Journal publish original research work that contributes significantly to further the scientific knowledge in engineering and Technology.
IRJET- Survey on Efficient Signal Processing Techniques for Speech EnhancementIRJET Journal
This document provides a survey of various speech enhancement techniques. It discusses five papers that propose different speech enhancement algorithms: 1) Discrete Tchebichef Transform and Discrete Krawtchouk Transform for removing noise using minimum mean square error. 2) Empirical mode decomposition and adaptive centre weighted average filtering that is effective for removing noise components. 3) Adaptive Wiener filtering that adapts the filter transfer function based on speech signal statistics. 4) Compressive sensing based speech enhancement that handles non-sparse noise. 5) Wavelet packet transform and non-negative matrix factorization to emphasize the speech components in each sub-band. The document also discusses speech enhancement using deep neural networks, empirical mode decomposition with Hurst exponent
Cochlear implant acoustic simulation model based on critical band filtersIAEME Publication
This document summarizes a research paper that proposes a new acoustic simulation model for cochlear implants based on critical band filters. The model uses a 16-channel critical band finite impulse response (CB-FIR) filter bank to analyze input speech signals. The filters decompose signals into different frequency bands corresponding to the human auditory system. Spectral energies are computed from the filter outputs to identify the most significant channels for stimulation. Temporal information is extracted through envelope detection. The model is tested on phonemes from the TIMIT speech database. Results show the model can provide stimulation to electrodes with minimal channel interaction by selecting only the most significant channels carrying speech information. The proposed model could help improve speech perception for cochlear implant users.
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
This document discusses using decision feedback equalization to enhance the performance of optical communication systems. It proposes using a fractionally spaced decision feedback equalizer (FSDFE) combined with activity detection guidance (ADG) and tap decoupling (TD) to improve the equalizer's effectiveness. The FSDFE replaces the symbol spaced feedback filter with a fractionally spaced feedback filter to enhance stability, steady-state error performance, and convergence rate. Adding ADG and TD further improves the steady-state error performance and convergence rate by detecting active taps in the channel impulse response. Simulation results show the FSDFE with ADG and TD offers superior performance to the FSDFE without these techniques, with improved compensation of amplitude distortion.
Application of Digital Signal Processing In Echo Cancellation: A SurveyEditor IJCATR
The advanced communications world is worried talking more naturally by using hands free this help the human being to talk
more confidently without holding any of the devices such as microphones or telephones. Acoustic echo cancellation and noise
cancellers are quite interesting nowadays because they are required in many applications such as speakerphones and audio/video
conferencing. This paper describes an alternative method of estimating signals corrupted by additive noise or interference. Acoustic
echo cancellation problem was discussed out of different noise cancellation techniques by concerning different parameters with their
comparative results .The results shown are using some specific algorithm
Acoustic fMRI noise reduction: a perceived loudness approachDimitri Vrehen
This document discusses a study that measured the subjective loudness of acoustic noise from fMRI scanners. The study recorded noise from three echo planar imaging sequences on a 3 Tesla MRI scanner. In a psychophysical experiment with 9 subjects, the perceived loudness of the fMRI noise did not increase linearly with sound pressure level. Noises with lower damping factors and frequencies in the 2.5-6kHz range of ear sensitivity were perceived as louder. EPI sequences with suppressed frequencies in the ear's most sensitive range and a highly impulsive nature distributed over longer times should reduce perceived loudness of fMRI acoustic noise.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Design and Implementation of Polyphase based Subband Adaptive Structure for N...Pratik Ghotkar
With the tremendous growth in the Digital Signal processing technology, there are many techniques available to remove noise from the speech signals which is used in the speech processing. Widely used LMS algorithm is modified with much advancement but still there are many limitations are introducing. This paper consist of a new approach i.e. subband adaptive processing for noise cancelation in the speech signals. Subband processing employs the multirate signal processing. The polyphase based subband adaptive implementation finds better results in term of MMSE , PSNR and processing time; also the synthesis filter bank is works on the lower data rate which reduces the computational Burdon as compare to the direct implementation of Subband adaptive filter. The normalized least mean squares (NLMS) algorithm is a class of adaptive filter used.
This study investigated how acute low-tone sensorineural hearing loss (ALHL) affects the perception of binaural beats (BBs). The researchers measured the frequency ranges where BBs were not perceived in 22 ALHL patients, 11 recovered ALHL patients, 5 high-tone hearing loss patients, and 14 normal controls. They found the frequency ranges were significantly wider in ALHL patients and narrowed with hearing recovery. The ranges did not differ between high-tone loss patients and controls, suggesting low-frequency damage affects BB perception by degrading frequency coding. Endolymphatic hydrops presence did not correlate with frequency range widths.
The document discusses various topics related to sound reinforcement systems including:
- The basic components of a sound wave and their definitions.
- Different types of input and output transducers used in sound systems.
- Key specifications and measurements used to evaluate audio equipment performance such as frequency response, signal level, and impedance.
- Concepts like loudness, equal loudness contours, and how they relate to human hearing.
- Different types of microphones, their characteristics, and common applications.
- Components and functions of a mixing console like preamps, auxiliary sends, and monitor mixes.
This document discusses various issues that may arise with cochlear implants and potential ways to address patient complaints through programming adjustments. It outlines common complaints such as sound quality issues, discomfort, loudness problems, and facial nerve stimulation. For each complaint, it lists potential programming parameters that could be modified, such as stimulus levels, filter settings, electrode configurations, and coding strategies. It emphasizes considering an acclimatization period and reassessing programming over time before making changes. It also discusses soft failures where performance declines despite normal integrity testing.
This document summarizes and compares different speech enhancement methods for noisy Punjabi speech at the phoneme level. It describes spectral subtraction, Wiener filtering, Kalman filtering, and RASTA methods. It also proposes a hybrid method using features from Wiener and Kalman filtering. Phonemes of Punjabi language and the methodology are explained. Results applying various noise types at different signal-to-noise ratios on phonemes show the proposed method produces the best enhancement compared to other methods.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
This document summarizes information about noise, sound level measurements, and the effects of noise on humans. It discusses how the decibel scale is used to measure sound intensity, how the loudness of sounds is quantified using phons and sones, and how long-term exposure to noise can cause temporary or permanent hearing loss. It also outlines Occupational Safety and Health Administration standards for permissible noise exposure limits in the workplace.
The document describes a study that measured the frequency response and cutoff frequencies of six cell phones for recording/playback and transmission. For recording/playback, the Apple smartphones had the highest cutoff frequencies around 15,000 Hz, followed by Android smartphones around 6,000-11,000 Hz, and conventional phones around 3,000-3,600 Hz. However, for transmission all phones had cutoff frequencies within 6% of each other near 3,400 Hz, the typical range for phone networks. The document provides background on human hearing, speech, telephone systems, frequency response analysis, and defines cutoff frequencies.
This document discusses voice modification techniques. It compares four models for voice modification: LPC, H/S, TD-PSOLA, and MBR-PSOLA. TD-PSOLA relies on pitch-synchronous overlap-add and modifies the speech signal in the time domain based on analysis and synthesis markings. The document implements a voice modification system using TD-PSOLA that can modify vocal tract, pitch, and time scale parameters to change voice quality, such as making a female voice sound more husky or nasal. Results show the algorithm can effectively modify input voices as desired.
Speech processing strategies for cochlear prostheses the past, present and fu...iaemedu
This document provides a review of speech processing strategies used in cochlear implants. It discusses early single-channel implants and limitations. It then describes the development of multi-channel implants and different types of speech processing strategies used, including waveform strategies like Compressed Analog and Continuous Interleaved Sampling, feature extraction strategies that code formant frequencies, and hybrid strategies. Finally, it summarizes commonly used strategies in current commercial cochlear implants and concludes that strategies providing spectral information allow better speech understanding than those extracting speech features alone.
1) The document proposes a single-port wireless transceiver circuit to enable affordable audio-visual communication for small enterprises, addressing the expensive costs of traditional videoconferencing.
2) A key challenge is reducing audio feedback known as the "Larsen effect" that occurs when a microphone picks up sound from its own speaker, causing an annoying acoustic noise.
3) The paper describes an experimental circuit built by the authors that uses a feedback control unit to monitor microphone and speaker levels and suppress the Larsen effect, allowing for clear audio-visual transmission over a single wireless port. Testing showed the acoustic echo was eliminated when this anti-Larsen effect circuit was employed.
The document discusses the evolution of the NAL fitting formula from a linear to a compression formula with three gain targets. The research showed that for gently sloping losses, similar gain was needed at 65dB inputs. NAL aims to equalize loudness across frequencies rather than normalize to unaided levels. This preserves speech spectrum loudness relationships and is theorized to maximize speech understanding. NAL gain targets are calculated based on phonemic information from the Articulation Index. The two key features of NAL are equalizing loudness across frequencies and providing less gain where loss is worst.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Computational Engineering Research(IJCER)ijceronline
International Journal of Computational Engineering Research(IJCER) is an intentional online Journal in English monthly publishing journal. This Journal publish original research work that contributes significantly to further the scientific knowledge in engineering and Technology.
IRJET- Survey on Efficient Signal Processing Techniques for Speech EnhancementIRJET Journal
This document provides a survey of various speech enhancement techniques. It discusses five papers that propose different speech enhancement algorithms: 1) Discrete Tchebichef Transform and Discrete Krawtchouk Transform for removing noise using minimum mean square error. 2) Empirical mode decomposition and adaptive centre weighted average filtering that is effective for removing noise components. 3) Adaptive Wiener filtering that adapts the filter transfer function based on speech signal statistics. 4) Compressive sensing based speech enhancement that handles non-sparse noise. 5) Wavelet packet transform and non-negative matrix factorization to emphasize the speech components in each sub-band. The document also discusses speech enhancement using deep neural networks, empirical mode decomposition with Hurst exponent
Cochlear implant acoustic simulation model based on critical band filtersIAEME Publication
This document summarizes a research paper that proposes a new acoustic simulation model for cochlear implants based on critical band filters. The model uses a 16-channel critical band finite impulse response (CB-FIR) filter bank to analyze input speech signals. The filters decompose signals into different frequency bands corresponding to the human auditory system. Spectral energies are computed from the filter outputs to identify the most significant channels for stimulation. Temporal information is extracted through envelope detection. The model is tested on phonemes from the TIMIT speech database. Results show the model can provide stimulation to electrodes with minimal channel interaction by selecting only the most significant channels carrying speech information. The proposed model could help improve speech perception for cochlear implant users.
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
This document discusses using decision feedback equalization to enhance the performance of optical communication systems. It proposes using a fractionally spaced decision feedback equalizer (FSDFE) combined with activity detection guidance (ADG) and tap decoupling (TD) to improve the equalizer's effectiveness. The FSDFE replaces the symbol spaced feedback filter with a fractionally spaced feedback filter to enhance stability, steady-state error performance, and convergence rate. Adding ADG and TD further improves the steady-state error performance and convergence rate by detecting active taps in the channel impulse response. Simulation results show the FSDFE with ADG and TD offers superior performance to the FSDFE without these techniques, with improved compensation of amplitude distortion.
Application of Digital Signal Processing In Echo Cancellation: A SurveyEditor IJCATR
The advanced communications world is worried talking more naturally by using hands free this help the human being to talk
more confidently without holding any of the devices such as microphones or telephones. Acoustic echo cancellation and noise
cancellers are quite interesting nowadays because they are required in many applications such as speakerphones and audio/video
conferencing. This paper describes an alternative method of estimating signals corrupted by additive noise or interference. Acoustic
echo cancellation problem was discussed out of different noise cancellation techniques by concerning different parameters with their
comparative results .The results shown are using some specific algorithm
Acoustic fMRI noise reduction: a perceived loudness approachDimitri Vrehen
This document discusses a study that measured the subjective loudness of acoustic noise from fMRI scanners. The study recorded noise from three echo planar imaging sequences on a 3 Tesla MRI scanner. In a psychophysical experiment with 9 subjects, the perceived loudness of the fMRI noise did not increase linearly with sound pressure level. Noises with lower damping factors and frequencies in the 2.5-6kHz range of ear sensitivity were perceived as louder. EPI sequences with suppressed frequencies in the ear's most sensitive range and a highly impulsive nature distributed over longer times should reduce perceived loudness of fMRI acoustic noise.
Novel Approach of Implementing Psychoacoustic model for MPEG-1 Audioinventy
Research Inventy : International Journal of Engineering and Science is published by the group of young academic and industrial researchers with 12 Issues per year. It is an online as well as print version open access journal that provides rapid publication (monthly) of articles in all areas of the subject such as: civil, mechanical, chemical, electronic and computer engineering as well as production and information technology. The Journal welcomes the submission of manuscripts that meet the general criteria of significance and scientific excellence. Papers will be published by rapid process within 20 days after acceptance and peer review process takes only 7 days. All articles published in Research Inventy will be peer-reviewed.
Design and Implementation of Polyphase based Subband Adaptive Structure for N...Pratik Ghotkar
With the tremendous growth in the Digital Signal processing technology, there are many techniques available to remove noise from the speech signals which is used in the speech processing. Widely used LMS algorithm is modified with much advancement but still there are many limitations are introducing. This paper consist of a new approach i.e. subband adaptive processing for noise cancelation in the speech signals. Subband processing employs the multirate signal processing. The polyphase based subband adaptive implementation finds better results in term of MMSE , PSNR and processing time; also the synthesis filter bank is works on the lower data rate which reduces the computational Burdon as compare to the direct implementation of Subband adaptive filter. The normalized least mean squares (NLMS) algorithm is a class of adaptive filter used.
Single Channel Speech Enhancement using Wiener Filter and Compressive Sensing IJECEIAES
The speech enhancement algorithms are utilized to overcome multiple limitation factors in recent applications such as mobile phone and communication channel. The challenges focus on corrupted speech solution between noise reduction and signal distortion. We used a modified Wiener filter and compressive sensing (CS) to investigate and evaluate the improvement of speech quality. This new method adapted noise estimation and Wiener filter gain function in which to increase weight amplitude spectrum and improve mitigation of interested signals. The CS is then applied using the gradient projection for sparse reconstruction (GPSR) technique as a study system to empirically investigate the interactive effects of the corrupted noise and obtain better perceptual improvement aspects to listener fatigue with noiseless reduction conditions. The proposed algorithm shows an enhancement in testing performance evaluation of objective assessment tests outperform compared to other conventional algorithms at various noise type conditions of 0, 5, 10, 15 dB SNRs. Therefore, the proposed algorithm significantly achieved the speech quality improvement and efficiently obtained higher performance resulting in better noise reduction compare to other conventional algorithms.
A NOVEL METHOD FOR OBTAINING A BETTER QUALITY SPEECH SIGNAL FOR COCHLEAR IMPL...acijjournal
Cochlear implant devices are known to exist since a long time. The purpose of the present work is to develop a speech algorithm for obtaining robust speech. In this paper, the technique of cochlear implant is first introduced, followed by discussions of some of the existing techniques available for obtaining speech. The next section introduces a new technique for obtaining robust speech. The key feature of this technique lies in the use of the advantages of an integrated approach involving the use of an estimation technique such as a kalman filter with non linear filter bank strategy, using Dual Resonance Non Linear(DRNL) and Single Side Band(SSB) Encoding method. A comparative study of the proposed method with the existing method indicates that the proposed method performs well compared to the existing method.
A Combined Sub-Band And Reconstructed Phase Space Approach To Phoneme Classif...April Smith
This paper presents a method for classifying phonemes that combines reconstructed phase space (RPS) representations with sub-band decomposition of speech signals. Experiments on the TIMIT database show that different phonological classes (vowels, fricatives, nasals, stops) are recognized with varying accuracy depending on the frequency sub-band. The results indicate filtering signals before embedding in RPS has potential to improve classification accuracy by exploiting differences in how well phonemes of different classes are represented in different frequency ranges. Combining classifications from multiple sub-bands may yield better performance than using the full-band signal alone.
Implementation of Digital Hearing AID for Sensory Neural Impairmentijtsrd
Hearing impairment is a chronicle disability affecting on people in world. The hearing aid is to amplify sound to overcome a hearing loss or impairment. The hearing aid picks up the sound signal with the microphone and amplifies all frequency sound signals but the sensory neural impairment person cannot hear particular frequency of sound in a noisy environment, since the auditory nerve is damaged. In this paper we are using MATLAB to design an adaptive filter for noise removal and filter banks for amplifies the particular frequency that a person with hearing loss can listen. Navya Bharathi K S | Bindu Shree C | Dr. V Udayashankara "Implementation of Digital Hearing AID for Sensory Neural Impairment" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-4 | Issue-4 , June 2020, URL: https://www.ijtsrd.com/papers/ijtsrd31133.pdf Paper Url :https://www.ijtsrd.com/engineering/bio-mechanicaland-biomedical-engineering/31133/implementation-of-digital-hearing-aid-for-sensory-neural-impairment/navya-bharathi-k-s
Active noise cancellation uses a microphone to measure ambient noise and generate an inverted "anti-noise" signal to destructively interfere with and cancel out the noise. It works best for low frequencies while passive noise control using insulation is more effective at higher frequencies. Adaptive noise cancellation algorithms like LMS analyze noise waveforms and generate inverted signals through transducers to reduce perceived noise levels. Noise-cancelling headphones apply this technique to improve listening and sleep on planes by offsetting engine noise.
HUFFMAN CODING ALGORITHM BASED ADAPTIVE NOISE CANCELLATIONIRJET Journal
This document presents a paper that proposes using Huffman coding and adaptive noise cancellation algorithms together to reduce noise while transmitting audio and visual signals. It begins with an abstract that discusses using data compression and algorithms to decrease the impact of background noise on recorded signals while maintaining the original undisturbed form of the signals. It then provides background on human hearing capabilities and digital audio signals. The document discusses existing noise cancellation systems and their limitations. It proposes a new framework that uses Huffman coding to create an adaptive code for each unique sound component, and builds a Huffman tree from those codes to map codes to probability. The proposed system is claimed to better remove signal transients and remaining noise artifacts compared to existing short-time Fourier transform approaches.
This document proposes a noise reduction method for audio signals based on an LMS adaptive filter. It segments the noisy audio signal into frames and uses an NLMS adaptive algorithm to estimate the filter coefficients and minimize the mean square error between the clean signal and filter output. Simulation results show the proposed method significantly reduces noise and improves the signal to noise ratio by adaptively filtering the noisy audio signal in the time domain. Analysis of the output signal variance indicates the noise level is substantially decreased compared to the original noisy signal.
The document discusses several medical applications of digital signal processing (DSP) including hearing aids, electroencephalograms (EEGs), and acquiring blood pressure signals. DSP techniques such as sampling, filtering, frequency analysis, and spectral estimation are used to process analog signals from the body, like brain waves or sound, into digital signals. This allows signals to be filtered and analyzed to extract clinically useful information for diagnosing conditions and monitoring patients.
Speech signal analysis for linear filter banks of different orderseSAT Journals
Abstract In speech signal processing using of filter banks is very important. The critical requirement is the sum of all the frequency responses of the band-pass filters of the filter bank i.e. composite frequency response be flat with linear phase. This paper deals with design and implementation of linear-phase FIR digital filters based filter bank flat with flat composite frequency response. The design is based on special properties of FIR filters by which excellent frequency response can be achieved. Keyword: Speech signal, FIR Filter, Composite frequency response.
The document discusses the importance of fully understanding a patient's residual hearing ability by measuring both their pure tone thresholds and loudness discomfort levels (LDLs). It emphasizes that LDLs should be measured at specific frequencies to define the patient's residual auditory area and dynamic range. The goal of hearing instrument fittings should be to amplify soft, average, and loud sounds within this residual area while avoiding exceeding the patient's LDLs. Modern digital devices allow for customization to meet these fitting goals without compromising gain or output. Understanding residual hearing is crucial for maximizing patient benefit from hearing instruments.
The document describes an active noise control system implemented using the LabVIEW platform to reduce noise in an acoustic waveguide. The system uses a feedforward FxLMS algorithm with additional filtering to suppress acoustic feedback between the control loudspeaker and reference microphone. Experimental results show the system achieved up to 50 dB reduction of a single tone noise and 12 dB reduction of multi-tone noise in a 1m duct. While effective at lower frequencies below 500-1000 Hz, active noise control has physical limitations for higher frequency noise cancellation due to the size of acoustic wavelengths involved.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
This document describes an automatic solar tracker designed using a microcontroller. It uses light dependent resistors as sensors to track the sun's position and maximize the efficiency of solar panels. The solar tracker operates in both normal and bad weather conditions. In normal conditions, the sensors detect sunlight and the solar panel tracks the sun. In bad weather when sensors cannot detect sunlight, the tracker rotates the panel in 3.75 degree increments every 15 minutes based on the earth's rotation. The tracker was tested and shown to successfully track the sun and increase in water temperature inside the solar cooker.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
This document describes a proposed smart home security system called AstroBell. The system uses a Wi-Fi enabled device with a push button, LCD screen, and USB camera located on the front door. It allows users to see and interact with visitors via their smartphone. Messages can be sent to the LCD screen and photos of visitors can be emailed. The system is powered by a cloud server that enables communication between the door device and smartphone. It aims to provide home security and visitor identification through internet of things technology in an early stage of development.
The Journal of MC Square Scientific Research is published by MC Square Publication on the monthly basis. It aims to publish original research papers devoted to wide areas in various disciplines of science and engineering and their applications in industry. This journal is basically devoted to interdisciplinary research in Science, Engineering and Technology, which can improve the technology being used in industry. The real-life problems involve multi-disciplinary knowledge, and thus strong inter-disciplinary approach is the need of the research.
Redefining brain tumor segmentation: a cutting-edge convolutional neural netw...IJECEIAES
Medical image analysis has witnessed significant advancements with deep learning techniques. In the domain of brain tumor segmentation, the ability to
precisely delineate tumor boundaries from magnetic resonance imaging (MRI)
scans holds profound implications for diagnosis. This study presents an ensemble convolutional neural network (CNN) with transfer learning, integrating
the state-of-the-art Deeplabv3+ architecture with the ResNet18 backbone. The
model is rigorously trained and evaluated, exhibiting remarkable performance
metrics, including an impressive global accuracy of 99.286%, a high-class accuracy of 82.191%, a mean intersection over union (IoU) of 79.900%, a weighted
IoU of 98.620%, and a Boundary F1 (BF) score of 83.303%. Notably, a detailed comparative analysis with existing methods showcases the superiority of
our proposed model. These findings underscore the model’s competence in precise brain tumor localization, underscoring its potential to revolutionize medical
image analysis and enhance healthcare outcomes. This research paves the way
for future exploration and optimization of advanced CNN models in medical
imaging, emphasizing addressing false positives and resource efficiency.
Presentation of IEEE Slovenia CIS (Computational Intelligence Society) Chapte...University of Maribor
Slides from talk presenting:
Aleš Zamuda: Presentation of IEEE Slovenia CIS (Computational Intelligence Society) Chapter and Networking.
Presentation at IcETRAN 2024 session:
"Inter-Society Networking Panel GRSS/MTT-S/CIS
Panel Session: Promoting Connection and Cooperation"
IEEE Slovenia GRSS
IEEE Serbia and Montenegro MTT-S
IEEE Slovenia CIS
11TH INTERNATIONAL CONFERENCE ON ELECTRICAL, ELECTRONIC AND COMPUTING ENGINEERING
3-6 June 2024, Niš, Serbia
DEEP LEARNING FOR SMART GRID INTRUSION DETECTION: A HYBRID CNN-LSTM-BASED MODELgerogepatton
As digital technology becomes more deeply embedded in power systems, protecting the communication
networks of Smart Grids (SG) has emerged as a critical concern. Distributed Network Protocol 3 (DNP3)
represents a multi-tiered application layer protocol extensively utilized in Supervisory Control and Data
Acquisition (SCADA)-based smart grids to facilitate real-time data gathering and control functionalities.
Robust Intrusion Detection Systems (IDS) are necessary for early threat detection and mitigation because
of the interconnection of these networks, which makes them vulnerable to a variety of cyberattacks. To
solve this issue, this paper develops a hybrid Deep Learning (DL) model specifically designed for intrusion
detection in smart grids. The proposed approach is a combination of the Convolutional Neural Network
(CNN) and the Long-Short-Term Memory algorithms (LSTM). We employed a recent intrusion detection
dataset (DNP3), which focuses on unauthorized commands and Denial of Service (DoS) cyberattacks, to
train and test our model. The results of our experiments show that our CNN-LSTM method is much better
at finding smart grid intrusions than other deep learning algorithms used for classification. In addition,
our proposed approach improves accuracy, precision, recall, and F1 score, achieving a high detection
accuracy rate of 99.50%.
A SYSTEMATIC RISK ASSESSMENT APPROACH FOR SECURING THE SMART IRRIGATION SYSTEMSIJNSA Journal
The smart irrigation system represents an innovative approach to optimize water usage in agricultural and landscaping practices. The integration of cutting-edge technologies, including sensors, actuators, and data analysis, empowers this system to provide accurate monitoring and control of irrigation processes by leveraging real-time environmental conditions. The main objective of a smart irrigation system is to optimize water efficiency, minimize expenses, and foster the adoption of sustainable water management methods. This paper conducts a systematic risk assessment by exploring the key components/assets and their functionalities in the smart irrigation system. The crucial role of sensors in gathering data on soil moisture, weather patterns, and plant well-being is emphasized in this system. These sensors enable intelligent decision-making in irrigation scheduling and water distribution, leading to enhanced water efficiency and sustainable water management practices. Actuators enable automated control of irrigation devices, ensuring precise and targeted water delivery to plants. Additionally, the paper addresses the potential threat and vulnerabilities associated with smart irrigation systems. It discusses limitations of the system, such as power constraints and computational capabilities, and calculates the potential security risks. The paper suggests possible risk treatment methods for effective secure system operation. In conclusion, the paper emphasizes the significant benefits of implementing smart irrigation systems, including improved water conservation, increased crop yield, and reduced environmental impact. Additionally, based on the security analysis conducted, the paper recommends the implementation of countermeasures and security approaches to address vulnerabilities and ensure the integrity and reliability of the system. By incorporating these measures, smart irrigation technology can revolutionize water management practices in agriculture, promoting sustainability, resource efficiency, and safeguarding against potential security threats.
Introduction- e - waste – definition - sources of e-waste– hazardous substances in e-waste - effects of e-waste on environment and human health- need for e-waste management– e-waste handling rules - waste minimization techniques for managing e-waste – recycling of e-waste - disposal treatment methods of e- waste – mechanism of extraction of precious metal from leaching solution-global Scenario of E-waste – E-waste in India- case studies.
International Conference on NLP, Artificial Intelligence, Machine Learning an...gerogepatton
International Conference on NLP, Artificial Intelligence, Machine Learning and Applications (NLAIM 2024) offers a premier global platform for exchanging insights and findings in the theory, methodology, and applications of NLP, Artificial Intelligence, Machine Learning, and their applications. The conference seeks substantial contributions across all key domains of NLP, Artificial Intelligence, Machine Learning, and their practical applications, aiming to foster both theoretical advancements and real-world implementations. With a focus on facilitating collaboration between researchers and practitioners from academia and industry, the conference serves as a nexus for sharing the latest developments in the field.
Understanding Inductive Bias in Machine LearningSUTEJAS
This presentation explores the concept of inductive bias in machine learning. It explains how algorithms come with built-in assumptions and preferences that guide the learning process. You'll learn about the different types of inductive bias and how they can impact the performance and generalizability of machine learning models.
The presentation also covers the positive and negative aspects of inductive bias, along with strategies for mitigating potential drawbacks. We'll explore examples of how bias manifests in algorithms like neural networks and decision trees.
By understanding inductive bias, you can gain valuable insights into how machine learning models work and make informed decisions when building and deploying them.
Batteries -Introduction – Types of Batteries – discharging and charging of battery - characteristics of battery –battery rating- various tests on battery- – Primary battery: silver button cell- Secondary battery :Ni-Cd battery-modern battery: lithium ion battery-maintenance of batteries-choices of batteries for electric vehicle applications.
Fuel Cells: Introduction- importance and classification of fuel cells - description, principle, components, applications of fuel cells: H2-O2 fuel cell, alkaline fuel cell, molten carbonate fuel cell and direct methanol fuel cells.
Electric vehicle and photovoltaic advanced roles in enhancing the financial p...IJECEIAES
Climate change's impact on the planet forced the United Nations and governments to promote green energies and electric transportation. The deployments of photovoltaic (PV) and electric vehicle (EV) systems gained stronger momentum due to their numerous advantages over fossil fuel types. The advantages go beyond sustainability to reach financial support and stability. The work in this paper introduces the hybrid system between PV and EV to support industrial and commercial plants. This paper covers the theoretical framework of the proposed hybrid system including the required equation to complete the cost analysis when PV and EV are present. In addition, the proposed design diagram which sets the priorities and requirements of the system is presented. The proposed approach allows setup to advance their power stability, especially during power outages. The presented information supports researchers and plant owners to complete the necessary analysis while promoting the deployment of clean energy. The result of a case study that represents a dairy milk farmer supports the theoretical works and highlights its advanced benefits to existing plants. The short return on investment of the proposed approach supports the paper's novelty approach for the sustainable electrical system. In addition, the proposed system allows for an isolated power setup without the need for a transmission line which enhances the safety of the electrical network
Electric vehicle and photovoltaic advanced roles in enhancing the financial p...
Ijmsr 2016-07
1. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
60
LOW-POWER HIGH-SPEED AMPLIFICATION USING
FILTERBANK FOR DIGITAL HEARING AIDS
C. Rajalakshmi
M.E Applied Electronics
Sri Balaji Chockalingam Engg., College
Arni, TN, India
K.Gunasekaran
Asso. Professor/ECE
Sri Balaji Chockalingam Engg., College
Arni, TN, India
Abstract - The ANSI S1.11 1/3-octave filter bank is suitable for digital hearing aids, but its
large group delay and high compu- tational complexity complicate matters considerably. This
study presents a 10-ms 18-band quasi-ANSI S1.11 1/3-octave filter bank for processing 24
kHz audio signals. We first discuss a filter order optimization algorithm to define the quasi-
ANSI filters. The group delay constraint of filters is limited to 10 ms. The proposed design
adopts an efficient prescription-fitting algorithm to reduce inter-band interference, enabling the
proposed quasi-ANSI filter bank to compensate any type of hearing loss (HL) using the
NAL-NL1 or HSE prescription formulas. Simulation results re- veal that the maximum
matching error in the prescriptions of the mild HL, moderate HL, and severe-to-profound HL is
less than 1.5 dB. This study also investigates the complexity-effective multirate IFIR quasi-
ANSI filter bank. For an 18-band digital hearing aid with a 24 kHz sampling rate, the proposed
architecture eliminates approximately 93% of the multiplications and up to 74% of the storage
elements, compared with a parallel FIR filters architec- ture. The proposed analysis filter bank
(AFB) was designed in UMC 90 nm CMOS high-VT technology, and on the basis of post-
layout simulations, it consumes 73 W. By voltage scaling (to 0.6 V), the simulation results
show that the power consumption decreases to 27 W, which is approximately 30% of that
consumed by the most energy-efficient AFB available in the literature for use in hearing aids.
Index Terms—Filter bank, hearing aid, low group delay.
1. Introduction
EARING loss [1]–[3] can be characterized as conduc- tive, sensorineural, and mixed
hearing loss. Conductive hearing loss means the sound is not conducted well through a
disordered outer or middle ear. Sensorineural hearing loss (SNHL) means the sensory
cells in the cochlea are absent or not functioning appropriately. If both conductive and
sensorineural losses are present, the result is mixed hearing loss. Conductive hearing loss can
be recovered after some adequate treatments, but most people with SNHL are fitted with
hearing aids. SNHL can degrade the functions of human ear in several different ways and
introduce phenomena such as a raised hearing threshold, de-creased and squeezed hearing range,
2. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
61
reduced temporal and spec- tral resolution, and the loss of noise tolerance [1]. These factors
make hearing aids more complex than simply amplifying sound
Audiologists usually identify and diagnose hearing loss with the pure tone audiogram (PTA)
test, which uses sinusoidal sig- nals over octave frequencies from 250 Hz to 8 kHz to
measure the minimum levels of sound (i.e., hearing thresholds). The re- sults of PTA test are
generally recorded on an audiogram. Fig. 1 demonstrates a typical example of moderate-to-
severe hearing loss. Fitting hearing aids usually requires a prescription formula. The widely
used NAL-NL1 [4], or the HSE for Chinese [5], gen- erates the ideal electro-acoustic
response (i.e., the gain-curve) of a hearing aid. The gain-curve specifies the insertion gain, or
the amplification, at each standard 1/3-octave frequency from 150 Hz to 8 kHz. The goal of
the NAL-NL1 is to maximize the speech intelligibility while maintaining the loudness of the
am- plified sound equal to, or less than, that perceived by people with normal hearing. The
NAL-NL1 produces different gain-curves for different input sound pressure levels (SPLs)
(e.g., 40, 50, 60, 65, 70, 80, and 90 dB). The right side of Fig. 1 illustrates the ex- ample
prescription of a 40 dB SPL input level.
Advanced hearing aids are currently battery-powered digital devices consisting of a
microphone, digital signal processing (DSP) circuit, and receiver (i.e., the loudspeaker) [1]–
[3]. The microphone and the receiver perform the transformation be- tween acoustic and
electrical signals. The DSP circuit performs sophisticated functions including the auditory
compensation algorithm to overcome the hearing loss, and noise reduction and feedback
cancellation to improve speech quality and intelligibility. The DSP circuit also uses adaptive
directional microphones and spectral shaping for speech enhancement. According to Kates
[3], a DSP block, performing entire set of DSP functions, typically consumes up to 61% of
the overall power budget of a digital hearing aid. One common approach to realize the
auditory compensation algorithm, which makes the sound audible for hearing-aid wearer, is
to employ an analysis filter bank (AFB) followed by sub-band amplifica- tion and multi-
channel wide dynamic-range control (WDRC) and an synthesis filter bank (SFB) [1]–[3], [6],
[14]. A low power Mandarin-specific hearing aid test chip was recently implemented in UMC
90 nm CMOS technology with High-VT standard cells [6]. The test chip contains an 18-band
filter bank and 3-channel WDRC auditory compensator and a multi-band noise reduction with
entropy enhanced voice activity detection (VAD). The power consumed by AFB is
approximately 27% of the total power in [6].
3. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
62
Fig. 1. Audiogram versus prescription formula plot for 40 dB SPL input level.
band [9]–[11], critical-band [12], symmetric-band [13], and 1/3- octave-band [14] filter banks.
A 7-band octave filter bank was designed in [9] and [10] using the interpolated FIR (IFIR)
filter technique. Lian and Wei [11] proposed an 8-band octave filter bank with the IFIR and
frequency-response masking (FRM) techniques to reduce the computational complexity.
Chong et al. designed a critical-band filter bank to match well human percep- tion [12].
However, the irregular property of the critical bands makes their implementation difficult. In
addition to [11], Wei and Lian proposed a 16-band symmetric filter bank [13] that
guarantees high frequency resolution at both high and low fre- quency regions but rather low
resolution near to . Kuo et al. recently proposed an efficient 18-band ANSI S1.11 1/3-
octave filter bank [14]. This 1/3-octave filter bank is suitable for both NAL-NL1 and HSE
prescription formulas for hearing aids.
To design a filter bank for hearing aids, the frequency re- sponse should match the
prescription as closely as possible. Suppose that the prescriptions by NAL-NL1 or HSE in
Fig. 1 are the target specification, and we evaluate the matching ca- pability of different
types of filter bank (Fig. 3). Further as- sume that filter banks in Fig. 3 possess 18 bands and
the pre- scription-fitting algorithm, described in Section II, is applied to minimize the
matching error. The uniform filter bank has equal-space sub-band bandwidth, which results in
a fixed fre- quency resolution. The lowest resolution in the low frequency region contributes the
maximum matching error, which is ap- proximately 8.4 dB. The symmetric filter bank, on the
other hand, has a rather low frequency resolution near . The max- imum matching error,
appearing in the middle frequency region, equals 6.2 dB. With matching to human hearing
characteristics, the critical-like filter bank reduces the maximum matching error to 3 dB. Finally,
the 18-band ANSI S1.11 1/3-octave filter bank achieves zero matching error because the
frequency sampling points of NAL-NL1 or HSE are the same as the central frequen- cies of
ANSI filter bank [15].
Filters usually cause delays in the datapath of the hearing aid. Although the 1/3-octave
filter bank has the best matching capability, it suffers from 78 ms delay for processing 24
kHz
4. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
63
Fig. 2. Different types of filter bank.
audio [14]. The delay of the 1/3-octave filter bank is still up to 27 ms if parallel minimum-
phase infinite-impulse response (IIR) filters are applied [14]. This is because the sharp transi-
tion bandwidth of the ANSI filter is defined in a low frequency region [15]. Except for ANSI
filter bank, the other filter banks in Fig. 3 have delays of approximately 10 ms. The matching
ca- pability of different filter banks obeys the acoustic uncertainty principle, which states that
the time-bandwidth product is con- stant. That is, if spectral resolution increases, temporal
resolu- tion decreases, and vice versa.
Hearing aids transmit signals into the ear canals through two different paths. One is the
directly received sound and the other is the sound processed by the hearing aid.
5. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
64
Fig. 3. Matching capability comparisons for different types of filter bank
Previous studies have investigated the acceptable delay introduced by the hearing aid. The
general requirement of less than 12 ms [2] prevents the loss of visual cues (un-synchronized)
with respect to hearing. Stone and Moore [16], [17] indicated that a delay of 20–30 ms can be
judged as objectionable for mild-to-moderate hearing loss. The popularity of open-canal (OC)
fitting hearing aids, which leave the ear canal much more open than traditional close-fitting
ear- molds, makes hearing aid delays even more concerning. In the OC fitting hearing aid,
more sounds would travel directly into the ear canal. A delay of approximately 10 ms might
create the comb filter effect [18], [19] (which will not be the case at most of frequencies) if
the direct path signal amplitude is comparable to the one produced by the hearing aid.
Using the high performance ANSI 1/3-octave filter bank, a relaxed-version with a low group
delay filter bank, called the quasi-ANSI filter bank, for the digital hearing aid is designed
and implemented. This study proposes a filter order optimiza- tion algorithm for developing
the FIR filters. The delay con- straint of each filter is limited to 10 ms. To reduce the
match error, this study also considers an efficient prescription fitting algorithm. Simulation
results show that the maximum matching error to various prescriptions of different types of
hearing loss is less than 1.5 dB. Moreover, a low complexity multirate IFIR filter bank
architecture is proposed. Compared with an 18-band parallel FIR filters, this design saves
approximately 93% of the multiplications and 74% of the storage elements. The proposed
analysis filter bank has also been implemented in UMC 90 nm CMOS technology with a high-
VT standard cell library. By pro- cessing 24 kHz audio, the chip consumes only 73 W. Applying
voltage scaling enables further energy savings. If the supply voltage decreases to 0.6 V, the
simulation result reveals that the power consumption of the proposed analysis filter bank equals
27 W, which is about 30% of that consumed by the most en-ergy-efficient AFB [14] available
in the literature design for the hearing aid.
The rest of this paper is organized as follows. Section II presents a low delay quasi-
ANSI S1.11 1/3-octave filter bank using a filter order optimization algorithm and an
efficient prescription-fitting algorithm to minimize the matching error.
6. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
65
Several simulation results in this section verify the effec-tiveness of the proposed filter
bank. Section III develops the low-complexity VLSI architecture of the proposed filter
bank by exploiting the IFIR and multirate signal processing tech- niques. Section IV
demonstrates the implementation result of the proposed filter bank. Finally, Section V
presents some concluding remarks.
2. Low Delay Filterbank Design
This section presents octave FIR filter bank as ζ.
A. Quasi-ANSI S1.11 1/3-Octave Filter Bank The ANSI S1.11 standard
[15] defines 3-class, 43 1/3-oc- tave bands covering the frequency range of
0–20 kHz. Each 1/3-octave band is specified by its midband frequency (or
cen- tral frequency) and bandwidth.
Fig. 4. (a) ANSI S1.11 class-2 filter specification [15] and, (b) parameters of the designed
filter.
7. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
66
Fig. 5. Quasi-ANSI S1.11 1/3-octave filter coefficient optimization algorithm.
Based on the good matching performance, this study de- signs a relaxed-version of
standard ANSI filters of constraint tap-length for digital hearing aids. Fig. 5 outlines the
proposed filter coefficient optimization algorithm, which contains two iter- ative design
procedures: one meets the 10 ms group delay constraint, and the other limits the relaxation in
the matching error. Note that an advanced noise reduction algorithm, such as the Siemens
SoundSmoothing noise reduction algorithm [20], contributes a nearly 1 ms group delay
[19]. Therefore, the constraint of 10 ms group delay of the filter bank is sufficient to meet the
general requirement of the hearing aid without loss of visual cues with respect to hearing [2].
Moreover, to design a filter bank for the hearing aid, the frequency response should match the
prescription as closely as possible. A 3 dB error performance is also a necessary constraint to
achieve the preferable compensation for each hearing loss pattern.
Note that expanding the transition bandwidth reduces the group delay of the designed filter
by (4) and (5).
8. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
67
TABLE I
Exploration Results of filter
The Minimize matching-error algorithm reduces the matching error caused by inter-band in-
terferences.
To evaluate the effectiveness of the proposed filter bank, this study uses audiograms from the
Independent Hearing Aid Infor- mation, a public service of Hearing Alliance of America [21].
These audiograms include mild hearing loss, moderate hearing loss, and severe-to-profound
hearing loss. These audiograms also appear in [11], but they considered fitting the audiograms
only, and not their prescriptions.
The audiogram in Fig. 7(a) depicts low frequency mild-to- moderate hearing loss and
mild high frequency hearing loss.
Fig. 6. Magnitude response comparison between (a) standard ANSI filters and (b) quasi-ANSI
S1.11 1/3-octave filters.
People with this type of hearing loss lose overall loudness be- cause most vowels cannot be
heard. Very close distance con- versations should be necessary. The maximum matching
error of the proposed filter bank is approximately 0.1 dB. The audio- gram in Fig. 7(b), like
9. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
68
that in Fig. 1, reveals moderate-to-severe hearing loss at middle to high frequency region,
which is the common type of hearing loss caused by aging. The sensitivity at low frequencies
is good enough to get some vowel information, helping the person realize that someone is
talking.
However, without consonants, they cannot easily distinguish between one word and
another. The maximum matching error of the pro- posed filter bank is approximately 0.4
dB, which is slightly worse than 0 dB, the standard ANSI filter bank, but much better than
the others in Fig. 3. The audiogram in Fig. 7(c) reveals severe-to-profound hearing loss at
middle to high frequency re- gion, which occurs commonly in older workers exposed to noisy
environments for prolonged periods. The maximum matching error of the proposed filter
bank is approximately 0.6 dB. Fi- nally, the audiogram in Fig. 7(d) shows severe flat
hearing loss at all frequencies, where the hearing thresholds are more than 70 dB.
Although this is a difficult case to compensate for, the maximum matching error is less than
1.5 dB, thus validating the effectiveness of the proposed filter bank.
3. Multirate Ifir Quasi-Ansi Filter Bank
This section presents the efficient VLSI architecture of the proposed filter bank by
exploiting the IFIR and multirate signal processing techniques.
Fig. 7. Matching results for different types of hearing loss: (a) mild to moderate hearing loss
in low frequencies, (b) hearing loss due to aging, (c) noise induced deafness, (d) severe to
profound flat hearing loss.
Fig. 8. (a) Illustrations of multirate IFIR implementation, and (b) noble iden- tity.
10. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
69
Although normal human ears are not sensitive to phase-delay, designing filter bank with exact
linear phase [11], [12], [14], of- fers some advantages regarding the development of advanced
binaural hearing aids, which not only target at compensating hearing losses, but also music
signals and sound localization for binaural hearing aids.
Fig. 10. Proposed 18-band multirate IFIR quasi-ANSI filter bank.
Fig. 11. Computation scheduling for the proposed filter bank.
4. Low-Power VLSI Implementation
One important issue in early stage of the system design is to decide the appropriate design
parameters among possible de- sign alternatives or design spaces. The design spaces
usually involve multiple metrics of interest, such as timing, resource usage, power, and cost. In
general, less functional units require higher clock rate and temporary storages or complicated
con- trol logic. Consider the silicon implementation in [14] as an ex- ample. By applying a
single multiply-and-accumulate (MAC) unit, standard ANSI analysis filter bank was
implemented in TSMC 130 nm CMOS technology and the chip operated at 6.13 MHz for
real-time processing of 24 kHz data.
11. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
70
It may be too high for hearing aid applications. And, the MAC unit occupies only
approximately 25% of the chip area and consumes approximately 30% of the total power
[14]. That is, the control logic and the storages are dominant, which may not be a good
architecture for low-power VLSI
A. Multi-MAC Architecture
Instead of single MAC unit, consider a set of 25 parallel mul- tipliers, which can
perform up to 49-tap linear-phase FIR fil- tering calculation in one cycle. With
25 multipliers, the first delay line requires 5 cycles to complete filtering
calculations for every sample, the second delay line requires 3 cycles for every two
samples, and the third delay line requires 21 cycles for every four samples. If data
are well scheduled, there will be no stall cycle and the hardware can operate at
288 kHz for real-time processing of 24 kHz audio. Otherwise, a higher clock rate
will be necessary.
1) Filter-Oriented RPA Algorithm: For simplicity, assume that within a clock cycle
there would be one, and only one, sub-filter with the right to access the set of 25
multipliers. The efficient data scheduling algorithm can be derived by modifying the RPA in
Fig. 11, called the filter-oriented RPA algorithm. Therefore, at most 12 cycles per sample are
required to accomplish the filtering operations. Note that the unused multipliers in each cycle
can be clock-gated for saving power.
The second row of Table VI shows the implementation result of the proposed quasi-ANSI
AFB, using filter-oriented RPA, in UMC 90 nm CMOS high-VT technology. Three 250 ms
input sequences were used for power estimation: a female voice, male voice, and random
signal. Synopsys PrimeTime suite and Nanosim were respectively applied to gate-level and
circuit-level simulations to evaluate the power performance. The clock rate of the proposed
quasi-ANSI AFB was 288 kHz and the power consumption was 91µw.
2) DelayLine-Oriented RPA Algorithm: The filter-oriented RPA is comprehensible;
however, the data fed into the set of 25 multipliers would switch over delay lines
frequently. This might consume extra dynamic power. To address this issue, par- tition the
set of 25 multipliers into three independent subsets, dedicated to three delay-lines (i.e., 9
multipliers for the first delay-line, 3 multipliers for the second, and 13 multipliers for the
third).
The third row of Table VI outlines the implementation result of the proposed AFB by
applying delayline-oriented RPA. The clock rate was 288 kHz and the power consumption was
84µw. Note that switching over delay lines infrequently reduces the dynamic power to 31µ w,
comparing 41µ w with the filter- oriented RPA.
B. Adder-Based FIR Architecture
Although the control logic is simple, the results in Section IV.A conclude that the
allocation of 25 multipliers seems to be an overdesign. One efficient method to reduce the
12. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
71
redundant operations is to apply multiple constant multipli- cations (MCMs) [28] or
common sub-expression elimination (CSE) [29] method. [29] An efficient multiplier-
less adder-based) quantization framework for FIR filters was re- cently proposed in [30],
which allows explicit tradeoffs between the hardware complexity and the quantization error
to facilitate FIR filter design exploration. Simulation results reveal that the adder-based
architecture saves approximately 43% redundant additions, compared with the direct
implementation for each sub-filter.
To achieve the same clock rate (i.e., 288 kHz), a chain of 45 adders are allocated. The
fourth row of Table VI shows the implementation results of an adder-based 18-band AFB,
which consumes 137 W. Both the chip area and power consumption are significantly worse
than that of multi-MAC cases. This is because that the adder-based architecture usually
accompanies an extreme increase in storage elements for temporary values [27]. Despite
rather limited arithmetic units, the control logic of the adder-based filter bank is overly
complicated, and requires many large multiplexers. This overrides the benefit of the re-
duced resource usage.
For a fair comparison, we have re-implemented the result in [14] using the same
CMOS technology (i.e., UMC 90 nm CMOS high-VT technology). The simulation results in
Table VI show that the single-MAC architecture of [14] consumes 102µ w.
C. The Optimized Low-Power Architecture
The implementation results in Table VI show that the op- timized hardware would be a
compromise design consisiting of fewer, but enough, parallel multipliers, limited storage, and
control logics. As described in Section III, the integral com- parison ratios regarding the
multiplicative complexity for three delay-lines are approximately 3 : 1 : 4, respectively. Because
of possessing the least complexity, it is necessary to allocate one MAC unit for the second
delay-line to serve filtering calcula- tions. To guarantee adequate computer power
preventing from stall or wait cycles, the number of MACs designated for the first and the third
delay line, respectively, will be 3 and 4. With 3 multipliers, 33 cycles are required to
complete filtering calcula- tions for the first delay-line. The second and the third delay-lines
require 52 and 125 cycles, respectively, to complete calculations with 1 and 4 multipliers,
respectively.
The system controller coordinates the data flow, according to the scheduling algorithm, and
handles the input interface. The register module contains the coefficient memory and the
data memory. The coefficient memory stores the 14 sub-filter coeffi- cients, while the data
memory maintains 3 separate delay-lines. The filter engine contains 3 independent sets (i.e.,
3, 1, and 4, respectively) of MAC units, dedicated for three delay-lines. The optimized 10 ms
18-band quasi-ANSI AFB has been implemented in UMC 90 nm CMOS high-VT standard
cell li- brary. The chip has an area of approximately 33274 (2-input NAND) gates and
operates at 792 kHz. For processing of 24 kHz audio, the power consumption is approximately
73 W, es- timated using three 250 ms input sequences: the female voice, male voice, and
random signal.
13. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
72
5. Conclusion
This study presents a low-delay, high-performance, and low-power filter bank design
for advanced digital hearing aids. The standard ANSI S1.11 1/3-octave bank is rarely
adopted in hearing aids because of its high computation complexity and rather large
group delay, even though it has the advantage of good match to human hearing
characteristics. This study proposes a 10-ms 18-band quasi-ANSI S1.11 1/3-octave filter
bank with a slight relaxation the ANSI specification. The computation complexity is 226
MACs. The storage complexity is 187 registers for delay-line, 506 coefficients, and 300
buffer registers to meet linear-phase requirements. The proposed AFB was implemented in
UMC 90 nm CMOS high-VT technology, and operated at 792 kHz for real-time processing
of 24 kHz audio and consumed approximately 73 W with V supply voltage.
The chip can also operate at a low voltage (0.6 V) without any performance degradation.
The contributions of this study include the following: (1) a systematic framework for
developing more appropriate quasi-ANSI specification of filters for hearing aids that are
more easily implementable and realizable, as Section II shows; (2) a thorough design
space exploration method that exploits multirate and IFIR techniques to construct a VLSI
architecture that significantly reduces multiplicative complexity of the filter bank without
increasing the latency unduly, as described in Section III; and (3) an efficient data scheduling
algorithm and appropriate hardware resource allocation for the small chip area and ultra-low
power implementation of the proposed filter bank, as Section IV shows. For business
considerations, the detailed specifications of modern hearing aids are beyond disclosure, and it
is difficult to compare them with the proposed filter bank. Nevertheless, we believe that, if
NAL-NL1 or HSE prescription formula is applied, the proposed design is superior.
REFERENCES
1. H. Dillon, Hearing Aids. New York: Thieme Medical Publisher, 2001.
2. J. Katz, Handbook of Clinical Audiology, 5th ed. New Yorl: Lippin- cott Williams &
Wilkins, 2001.
3. J. M. Kates, Digital Hearing Aids. : Plural Publishing, 2008.
4. D. Byrne, H. Dillon, T. Ching, R. Katsch, and G. Keidser, ―NAL-NL1 procedure for fitting
nonlinear hearing aids: characteristics and com- parisons with other procedures,‖ J. Amer.
Acad. Audiol., vol. 12, no.
5. J. H. Chang, K. S. Tsai, P. C. Li, and S. T. Young, ―Computer-Aided simulation of multi-
channel WDRC hearing aids,‖ presented at the Proc. 17th Ann. Convention Expo Amer.
Acad. Audiology, Wash- ington, DC, 2005.
6. C.-W. Wei et al., ―A low-power mandarin-specific hearing aid chip,‖ in Proc. IEEE Asian
Solid-State Circuits Conference, Beijing, China, 2010.
7. R. Brennan and T. Schneider, ―A flexible filter bank structure for ex- tensive signal
manipulations in digital hearing aids,‖ in Proc. IEEE Int. Symp. Circuits Syst., 1998, pp. 569–
572.
14. International Journal of MC Square Scientific Research Vol.8, No.1 Nov 2016
73
8. T. Lunner and J. Hellgren, ―A digital filterbank hearing aid—Design, implementation and
evaluation,‖ in Proc. ICASSP Conf., 1991, pp.
9. L. S. Nielsen and J. Sparso, ―Designing asynchronous circuits for low power: An IFIR filter
bank for a digital hearing aid,‖ Proc. IEEE, vol.
10. Y. Lian and Y. Wei, ―A computationally efficient nonuniform FIR dig- ital filter bank for
hearing aids,‖ IEEE Tran. Circuits Syst. I, Reg. Pa- pers, vol. 52, no. 12, pp. 2754–2762,
Dec. 2005.
11. ] K. S. Chong, B. H. Gwee, and J. S. Chang, ―A 16-channel low-power nonuniform spaced
filter bank core for digital hearing aid,‖ IEEE Tran. Circuits Syst. I, Reg. Papers, vol. 53, no.
9, pp. 853–857, Sep. 2006.
12. Y. Wei and Y. Lian, ―A 16-band nonuniform FIR digital filterbank for hearing aid,‖ in Proc.
IEEE Biomed. Circuits Syst. Conf., 2006, pp.
13. Y. T. Kuo, T. J. Lin, Y. T. Li, and C. W. Liu, ―Design & implementation of low-power ANSI
S1.11 filter bank for digital hearing aids,‖ IEEE Tran. Circuits Syst. I, Reg. Papers, vol. 57,
no. 7, pp. 1684–1696, Jul. 2010.
14. Specification for Octave-Band and Fractional-Octave-Band Analog and Digital Filters,
ANSI Standard S1.11-2004.
15. M. A. Stone, B. C. J. Moore, K. Meisenbacher, and R. Derleth, ―Tol- erable hearing-aid
delays V—Estimation of limits for open canal fit- tings,‖ Ear and Hearing, vol. 29, no. 4,
pp. 601–617, 2008.
16. K.-C. Chang, Y.-T. Kuo, and C.-W. Liu, ―Low-complexity dynamic range compression for
digital hearing aids,‖ IEEE Tran. Circuits Syst., submitted for publication.
17. M. Mehendale and S. D. Sherlekar, VLSI Synthesis of DSP Ker- nels—Algorithmic and
Architectural Transformations. Norwell, MA: Kluwer, 2001.
18. P. P. Vaidyanathan, Multirate Systems and Filter Banks. Englewood 1993.