A Binaural Cochlear Implant Sound Coding Strategy Inspired by the Contralateral Medial Olivocochlear Reflex

PubMed Central

Eustaquio-Martín, Almudena; Stohl, Joshua S.; Wolford, Robert D.; Schatzer, Reinhold; Wilson, Blake S.

2016-01-01

Objectives: In natural hearing, cochlear mechanical compression is dynamically adjusted via the efferent medial olivocochlear reflex (MOCR). These adjustments probably help understanding speech in noisy environments and are not available to the users of current cochlear implants (CIs). The aims of the present study are to: (1) present a binaural CI sound processing strategy inspired by the control of cochlear compression provided by the contralateral MOCR in natural hearing; and (2) assess the benefits of the new strategy for understanding speech presented in competition with steady noise with a speech-like spectrum in various spatial configurations of the speech and noise sources. Design: Pairs of CI sound processors (one per ear) were constructed to mimic or not mimic the effects of the contralateral MOCR on compression. For the nonmimicking condition (standard strategy or STD), the two processors in a pair functioned similarly to standard clinical processors (i.e., with fixed back-end compression and independently of each other). When configured to mimic the effects of the MOCR (MOC strategy), the two processors communicated with each other and the amount of back-end compression in a given frequency channel of each processor in the pair decreased/increased dynamically (so that output levels dropped/increased) with increases/decreases in the output energy from the corresponding frequency channel in the contralateral processor. Speech reception thresholds in speech-shaped noise were measured for 3 bilateral CI users and 2 single-sided deaf unilateral CI users. Thresholds were compared for the STD and MOC strategies in unilateral and bilateral listening conditions and for three spatial configurations of the speech and noise sources in simulated free-field conditions: speech and noise sources colocated in front of the listener, speech on the left ear with noise in front of the listener, and speech on the left ear with noise on the right ear. In both bilateral and unilateral listening, the electrical stimulus delivered to the test ear(s) was always calculated as if the listeners were wearing bilateral processors. Results: In both unilateral and bilateral listening conditions, mean speech reception thresholds were comparable with the two strategies for colocated speech and noise sources, but were at least 2 dB lower (better) with the MOC than with the STD strategy for spatially separated speech and noise sources. In unilateral listening conditions, mean thresholds improved with increasing the spatial separation between the speech and noise sources regardless of the strategy but the improvement was significantly greater with the MOC strategy. In bilateral listening conditions, thresholds improved significantly with increasing the speech-noise spatial separation only with the MOC strategy. Conclusions: The MOC strategy (1) significantly improved the intelligibility of speech presented in competition with a spatially separated noise source, both in unilateral and bilateral listening conditions; (2) produced significant spatial release from masking in bilateral listening conditions, something that did not occur with fixed compression; and (3) enhanced spatial release from masking in unilateral listening conditions. The MOC strategy as implemented here, or a modified version of it, may be usefully applied in CIs and in hearing aids. PMID:26862711

A Binaural Cochlear Implant Sound Coding Strategy Inspired by the Contralateral Medial Olivocochlear Reflex.

PubMed

Lopez-Poveda, Enrique A; Eustaquio-Martín, Almudena; Stohl, Joshua S; Wolford, Robert D; Schatzer, Reinhold; Wilson, Blake S

2016-01-01

In natural hearing, cochlear mechanical compression is dynamically adjusted via the efferent medial olivocochlear reflex (MOCR). These adjustments probably help understanding speech in noisy environments and are not available to the users of current cochlear implants (CIs). The aims of the present study are to: (1) present a binaural CI sound processing strategy inspired by the control of cochlear compression provided by the contralateral MOCR in natural hearing; and (2) assess the benefits of the new strategy for understanding speech presented in competition with steady noise with a speech-like spectrum in various spatial configurations of the speech and noise sources. Pairs of CI sound processors (one per ear) were constructed to mimic or not mimic the effects of the contralateral MOCR on compression. For the nonmimicking condition (standard strategy or STD), the two processors in a pair functioned similarly to standard clinical processors (i.e., with fixed back-end compression and independently of each other). When configured to mimic the effects of the MOCR (MOC strategy), the two processors communicated with each other and the amount of back-end compression in a given frequency channel of each processor in the pair decreased/increased dynamically (so that output levels dropped/increased) with increases/decreases in the output energy from the corresponding frequency channel in the contralateral processor. Speech reception thresholds in speech-shaped noise were measured for 3 bilateral CI users and 2 single-sided deaf unilateral CI users. Thresholds were compared for the STD and MOC strategies in unilateral and bilateral listening conditions and for three spatial configurations of the speech and noise sources in simulated free-field conditions: speech and noise sources colocated in front of the listener, speech on the left ear with noise in front of the listener, and speech on the left ear with noise on the right ear. In both bilateral and unilateral listening, the electrical stimulus delivered to the test ear(s) was always calculated as if the listeners were wearing bilateral processors. In both unilateral and bilateral listening conditions, mean speech reception thresholds were comparable with the two strategies for colocated speech and noise sources, but were at least 2 dB lower (better) with the MOC than with the STD strategy for spatially separated speech and noise sources. In unilateral listening conditions, mean thresholds improved with increasing the spatial separation between the speech and noise sources regardless of the strategy but the improvement was significantly greater with the MOC strategy. In bilateral listening conditions, thresholds improved significantly with increasing the speech-noise spatial separation only with the MOC strategy. The MOC strategy (1) significantly improved the intelligibility of speech presented in competition with a spatially separated noise source, both in unilateral and bilateral listening conditions; (2) produced significant spatial release from masking in bilateral listening conditions, something that did not occur with fixed compression; and (3) enhanced spatial release from masking in unilateral listening conditions. The MOC strategy as implemented here, or a modified version of it, may be usefully applied in CIs and in hearing aids.

Speech Intelligibility in Various Noise Conditions with the Nucleus® 5 CP810 Sound Processor.

PubMed

Dillier, Norbert; Lai, Wai Kong

2015-06-11

The Nucleus(®) 5 System Sound Processor (CP810, Cochlear™, Macquarie University, NSW, Australia) contains two omnidirectional microphones. They can be configured as a fixed directional microphone combination (called Zoom) or as an adaptive beamformer (called Beam), which adjusts the directivity continuously to maximally reduce the interfering noise. Initial evaluation studies with the CP810 had compared performance and usability of the new processor in comparison with the Freedom™ Sound Processor (Cochlear™) for speech in quiet and noise for a subset of the processing options. This study compares the two processing options suggested to be used in noisy environments, Zoom and Beam, for various sound field conditions using a standardized speech in noise matrix test (Oldenburg sentences test). Nine German-speaking subjects who previously had been using the Freedom speech processor and subsequently were upgraded to the CP810 device participated in this series of additional evaluation tests. The speech reception threshold (SRT for 50% speech intelligibility in noise) was determined using sentences presented via loudspeaker at 65 dB SPL in front of the listener and noise presented either via the same loudspeaker (S0N0) or at 90 degrees at either the ear with the sound processor (S0NCI+) or the opposite unaided ear (S0NCI-). The fourth noise condition consisted of three uncorrelated noise sources placed at 90, 180 and 270 degrees. The noise level was adjusted through an adaptive procedure to yield a signal to noise ratio where 50% of the words in the sentences were correctly understood. In spatially separated speech and noise conditions both Zoom and Beam could improve the SRT significantly. For single noise sources, either ipsilateral or contralateral to the cochlear implant sound processor, average improvements with Beam of 12.9 and 7.9 dB in SRT were found. The average SRT of -8 dB for Beam in the diffuse noise condition (uncorrelated noise from both sides and back) is truly remarkable and comparable to the performance of normal hearing listeners in the same test environment. The static directivity (Zoom) option in the diffuse noise condition still provides a significant benefit of 5.9 dB in comparison with the standard omnidirectional microphone setting. These results indicate that CI recipients may improve their speech recognition in noisy environments significantly using these directional microphone-processing options.

Clinical Validation of a Sound Processor Upgrade in Direct Acoustic Cochlear Implant Subjects

PubMed Central

Kludt, Eugen; D’hondt, Christiane; Lenarz, Thomas; Maier, Hannes

2017-01-01

Objective: The objectives of the investigation were to evaluate the effect of a sound processor upgrade on the speech reception threshold in noise and to collect long-term safety and efficacy data after 2½ to 5 years of device use of direct acoustic cochlear implant (DACI) recipients. Study Design: The study was designed as a mono-centric, prospective clinical trial. Setting: Tertiary referral center. Patients: Fifteen patients implanted with a direct acoustic cochlear implant. Intervention: Upgrade with a newer generation of sound processor. Main Outcome Measures: Speech recognition test in quiet and in noise, pure tone thresholds, subject-reported outcome measures. Results: The speech recognition in quiet and in noise is superior after the sound processor upgrade and stable after long-term use of the direct acoustic cochlear implant. The bone conduction thresholds did not decrease significantly after long-term high level stimulation. Conclusions: The new sound processor for the DACI system provides significant benefits for DACI users for speech recognition in both quiet and noise. Especially the noise program with the use of directional microphones (Zoom) allows DACI patients to have much less difficulty when having conversations in noisy environments. Furthermore, the study confirms that the benefits of the sound processor upgrade are available to the DACI recipients even after several years of experience with a legacy sound processor. Finally, our study demonstrates that the DACI system is a safe and effective long-term therapy. PMID:28406848

A phone-assistive device based on Bluetooth technology for cochlear implant users.

PubMed

Qian, Haifeng; Loizou, Philipos C; Dorman, Michael F

2003-09-01

Hearing-impaired people, and particularly hearing-aid and cochlear-implant users, often have difficulty communicating over the telephone. The intelligibility of telephone speech is considerably lower than the intelligibility of face-to-face speech. This is partly because of lack of visual cues, limited telephone bandwidth, and background noise. In addition, cellphones may cause interference with the hearing aid or cochlear implant. To address these problems that hearing-impaired people experience with telephones, this paper proposes a wireless phone adapter that can be used to route the audio signal directly to the hearing aid or cochlear implant processor. This adapter is based on Bluetooth technology. The favorable features of this new wireless technology make the adapter superior to traditional assistive listening devices. A hardware prototype was built and software programs were written to implement the headset profile in the Bluetooth specification. Three cochlear implant users were tested with the proposed phone-adapter and reported good speech quality.

Speech understanding in noise with the Roger Pen, Naida CI Q70 processor, and integrated Roger 17 receiver in a multi-talker network.

PubMed

De Ceulaer, Geert; Bestel, Julie; Mülder, Hans E; Goldbeck, Felix; de Varebeke, Sebastien Pierre Janssens; Govaerts, Paul J

2016-05-01

Roger is a digital adaptive multi-channel remote microphone technology that wirelessly transmits a speaker's voice directly to a hearing instrument or cochlear implant sound processor. Frequency hopping between channels, in combination with repeated broadcast, avoids interference issues that have limited earlier generation FM systems. This study evaluated the benefit of the Roger Pen transmitter microphone in a multiple talker network (MTN) for cochlear implant users in a simulated noisy conversation setting. Twelve post-lingually deafened adult Advanced Bionics CII/HiRes 90K recipients were recruited. Subjects used a Naida CI Q70 processor with integrated Roger 17 receiver. The test environment simulated four people having a meal in a noisy restaurant, one the CI user (listener), and three companions (talkers) talking non-simultaneously in a diffuse field of multi-talker babble. Speech reception thresholds (SRTs) were determined without the Roger Pen, with one Roger Pen, and with three Roger Pens in an MTN. Using three Roger Pens in an MTN improved the SRT by 14.8 dB over using no Roger Pen, and by 13.1 dB over using a single Roger Pen (p < 0.0001). The Roger Pen in an MTN provided statistically and clinically significant improvement in speech perception in noise for Advanced Bionics cochlear implant recipients. The integrated Roger 17 receiver made it easy for users of the Naida CI Q70 processor to take advantage of the Roger system. The listening advantage and ease of use should encourage more clinicians to recommend and fit Roger in adult cochlear implant patients.

A novel speech-processing strategy incorporating tonal information for cochlear implants.

PubMed

Lan, N; Nie, K B; Gao, S K; Zeng, F G

2004-05-01

Good performance in cochlear implant users depends in large part on the ability of a speech processor to effectively decompose speech signals into multiple channels of narrow-band electrical pulses for stimulation of the auditory nerve. Speech processors that extract only envelopes of the narrow-band signals (e.g., the continuous interleaved sampling (CIS) processor) may not provide sufficient information to encode the tonal cues in languages such as Chinese. To improve the performance in cochlear implant users who speak tonal language, we proposed and developed a novel speech-processing strategy, which extracted both the envelopes of the narrow-band signals and the fundamental frequency (F0) of the speech signal, and used them to modulate both the amplitude and the frequency of the electrical pulses delivered to stimulation electrodes. We developed an algorithm to extract the fundatmental frequency and identified the general patterns of pitch variations of four typical tones in Chinese speech. The effectiveness of the extraction algorithm was verified with an artificial neural network that recognized the tonal patterns from the extracted F0 information. We then compared the novel strategy with the envelope-extraction CIS strategy in human subjects with normal hearing. The novel strategy produced significant improvement in perception of Chinese tones, phrases, and sentences. This novel processor with dynamic modulation of both frequency and amplitude is encouraging for the design of a cochlear implant device for sensorineurally deaf patients who speak tonal languages.

Perceptions of parents and speech and language therapists on the effects of paediatric cochlear implantation and habilitation and education following it.

PubMed

Huttunen, Kerttu; Välimaa, Taina

2012-01-01

During the process of implantation, parents may have rather heterogeneous expectations and concerns about their child's development and the functioning of habilitation and education services. Their views on habilitation and education are important for building family-centred practices. We explored the perceptions of parents and speech and language therapists (SLTs) on the effects of implantation on the child and the family and on the quality of services provided. Their views were also compared. Parents and SLTs of 18 children filled out questionnaires containing open- and closed-ended questions at 6 months and annually 1-5 years after activation of the implant. Their responses were analysed mainly using data-based inductive content analysis. Positive experiences outnumbered negative ones in the responses of both the parents and the SLTs surveyed. The parents were particularly satisfied with the improvement in communication and expanded social life in the family. These were the most prevalent themes also raised by the SLTs. The parents were also satisfied with the organization and content of habilitation. Most of the negative experiences were related to arrangement of hospital visits and the usability and maintenance of speech processor technology. Some children did not receive enough speech and language therapy, and some of the parents were dissatisfied with educational services. The habilitation process had generally required parental efforts at an expected level. However, parents with a child with at least one concomitant problem experienced habilitation as more stressful than did other parents. Parents and SLTs had more positive than negative experiences with implantation. As the usability and maintenance of speech processor technology were often compromised, we urge implant centres to ensure sufficient personnel for technical maintenance. It is also important to promote services by providing enough information and parental support. © 2011 Royal College of Speech & Language Therapists.

Electromagnetic versus electrical coupling of personal frequency modulation (FM) receivers to cochlear implant sound processors.

PubMed

Schafer, Erin C; Romine, Denise; Musgrave, Elizabeth; Momin, Sadaf; Huynh, Christy

2013-01-01

Previous research has suggested that electrically coupled frequency modulation (FM) systems substantially improved speech-recognition performance in noise in individuals with cochlear implants (CIs). However, there is limited evidence to support the use of electromagnetically coupled (neck loop) FM receivers with contemporary CI sound processors containing telecoils. The primary goal of this study was to compare speech-recognition performance in noise and subjective ratings of adolescents and adults using one of three contemporary CI sound processors coupled to electromagnetically and electrically coupled FM receivers from Oticon. A repeated-measures design was used to compare speech-recognition performance in noise and subjective ratings without and with the FM systems across three test sessions (Experiment 1) and to compare performance at different FM-gain settings (Experiment 2). Descriptive statistics were used in Experiment 3 to describe output differences measured through a CI sound processor. Experiment 1 included nine adolescents or adults with unilateral or bilateral Advanced Bionics Harmony (n = 3), Cochlear Nucleus 5 (n = 3), and MED-EL OPUS 2 (n = 3) CI sound processors. In Experiment 2, seven of the original nine participants were tested. In Experiment 3, electroacoustic output was measured from a Nucleus 5 sound processor when coupled to the electromagnetically coupled Oticon Arc neck loop and electrically coupled Oticon R2. In Experiment 1, participants completed a field trial with each FM receiver and three test sessions that included speech-recognition performance in noise and a subjective rating scale. In Experiment 2, participants were tested in three receiver-gain conditions. Results in both experiments were analyzed using repeated-measures analysis of variance. Experiment 3 involved electroacoustic-test measures to determine the monitor-earphone output of the CI alone and CI coupled to the two FM receivers. The results in Experiment 1 suggested that both FM receivers provided significantly better speech-recognition performance in noise than the CI alone; however, the electromagnetically coupled receiver provided significantly better speech-recognition performance in noise and better ratings in some situations than the electrically coupled receiver when set to the same gain. In Experiment 2, the primary analysis suggested significantly better speech-recognition performance in noise for the neck-loop versus electrically coupled receiver, but a second analysis, using the best performance across gain settings for each device, revealed no significant differences between the two FM receivers. Experiment 3 revealed monitor-earphone output differences in the Nucleus 5 sound processor for the two FM receivers when set to the +8 setting used in Experiment 1 but equal output when the electrically coupled device was set to a +16 gain setting and the electromagnetically coupled device was set to the +8 gain setting. Individuals with contemporary sound processors may show more favorable speech-recognition performance in noise electromagnetically coupled FM systems (i.e., Oticon Arc), which is most likely related to the input processing and signal processing pathway within the CI sound processor for direct input versus telecoil input. Further research is warranted to replicate these findings with a larger sample size and to develop and validate a more objective approach to fitting FM systems to CI sound processors. American Academy of Audiology.

The development of the Nucleus Freedom Cochlear implant system.

PubMed

Patrick, James F; Busby, Peter A; Gibson, Peter J

2006-12-01

Cochlear Limited (Cochlear) released the fourth-generation cochlear implant system, Nucleus Freedom, in 2005. Freedom is based on 25 years of experience in cochlear implant research and development and incorporates advances in medicine, implantable materials, electronic technology, and sound coding. This article presents the development of Cochlear's implant systems, with an overview of the first 3 generations, and details of the Freedom system: the CI24RE receiver-stimulator, the Contour Advance electrode, the modular Freedom processor, the available speech coding strategies, the input processing options of Smart Sound to improve the signal before coding as electrical signals, and the programming software. Preliminary results from multicenter studies with the Freedom system are reported, demonstrating better levels of performance compared with the previous systems. The final section presents the most recent implant reliability data, with the early findings at 18 months showing improved reliability of the Freedom implant compared with the earlier Nucleus 3 System. Also reported are some of the findings of Cochlear's collaborative research programs to improve recipient outcomes. Included are studies showing the benefits from bilateral implants, electroacoustic stimulation using an ipsilateral and/or contralateral hearing aid, advanced speech coding, and streamlined speech processor programming.

"Once upon a Time There Was a Mouse": Children's Technology-Mediated Storytelling in Preschool Class

ERIC Educational Resources Information Center

Skantz Åberg, Ewa; Lantz-Andersson, Annika; Pramling, Niklas

2014-01-01

With the current expansion of digital tools, the media used for narration is changing, challenging traditional literacies in educational settings. The present study explores what kind of activities emerge when six-year-old children in a preschool class write a digital story, using a word processor and speech-synthesised feedback computer software.…

Use of amplitude modulation cues recovered from frequency modulation for cochlear implant users when original speech cues are severely degraded.

PubMed

Won, Jong Ho; Shim, Hyun Joon; Lorenzi, Christian; Rubinstein, Jay T

2014-06-01

Won et al. (J Acoust Soc Am 132:1113-1119, 2012) reported that cochlear implant (CI) speech processors generate amplitude-modulation (AM) cues recovered from broadband speech frequency modulation (FM) and that CI users can use these cues for speech identification in quiet. The present study was designed to extend this finding for a wide range of listening conditions, where the original speech cues were severely degraded by manipulating either the acoustic signals or the speech processor. The manipulation of the acoustic signals included the presentation of background noise, simulation of reverberation, and amplitude compression. The manipulation of the speech processor included changing the input dynamic range and the number of channels. For each of these conditions, multiple levels of speech degradation were tested. Speech identification was measured for CI users and compared for stimuli having both AM and FM information (intact condition) or FM information only (FM condition). Each manipulation degraded speech identification performance for both intact and FM conditions. Performance for the intact and FM conditions became similar for stimuli having the most severe degradations. Identification performance generally overlapped for the intact and FM conditions. Moreover, identification performance for the FM condition was better than chance performance even at the maximum level of distortion. Finally, significant correlations were found between speech identification scores for the intact and FM conditions. Altogether, these results suggest that despite poor frequency selectivity, CI users can make efficient use of AM cues recovered from speech FM in difficult listening situations.

The design of an adaptive predictive coder using a single-chip digital signal processor

NASA Astrophysics Data System (ADS)

Randolph, M. A.

1985-01-01

A speech coding processor architecture design study has been performed in which Texas Instruments TMS32010 has been selected from among three commercially available digital signal processing integrated circuits and evaluated in an implementation study of real-time Adaptive Predictive Coding (APC). The TMS32010 has been compared with AR&T Bell Laboratories DSP I and Nippon Electric Co. PD7720 and was found to be most suitable for a single chip implementation of APC. A preliminary design system based on TMS32010 has been performed, and several of the hardware and software design issues are discussed. Particular attention was paid to the design of an external memory controller which permits rapid sequential access of external RAM. As a result, it has been determined that a compact hardware implementation of the APC algorithm is feasible based of the TSM32010. Originator-supplied keywords include: vocoders, speech compression, adaptive predictive coding, digital signal processing microcomputers, speech processor architectures, and special purpose processor.

The use of cochlear's SCAN and wireless microphones to improve speech understanding in noise with the Nucleus6® CP900 processor.

PubMed

De Ceulaer, Geert; Pascoal, David; Vanpoucke, Filiep; Govaerts, Paul J

2017-11-01

The newest Nucleus CI processor, the CP900, has two new options to improve speech-in-noise perception: (1) use of an adaptive directional microphone (SCAN mode) and (2) wireless connection to MiniMic1 and MiniMic2 wireless remote microphones. An analysis was made of the absolute and relative benefits of these technologies in a real-world mimicking test situation. Speech perception was tested using an adaptive speech-in-noise test (sentences-in-babble noise). In session A, SRTs were measured in three conditions: (1) Clinical Map, (2) SCAN and (3) MiniMic1. Each was assessed for three distances between speakers and CI recipient: 1 m, 2 m and 3 m. In session B, the benefit of the use of MiniMic2 was compared to benefit of MiniMic1 at 3 m. A group of 13 adult CP900 recipients participated. SCAN and MiniMic1 improved performance compared to the standard microphone with a median improvement in SRT of 2.7-3.9 dB for SCAN at 1 m and 3 m, respectively, and 4.7-10.9 dB for the MiniMic1. MiniMic1 improvements were significant. MiniMic2 showed an improvement in SRT of 22.2 dB compared to 10.0 dB for MiniMic1 (3 m). Digital wireless transmission systems (i.e. MiniMic) offer a statistically and clinically significant improvement in speech perception in challenging, realistic listening conditions.

Electroacoustic verification of frequency modulation systems in cochlear implant users.

PubMed

Fidêncio, Vanessa Luisa Destro; Jacob, Regina Tangerino de Souza; Tanamati, Liége Franzini; Bucuvic, Érika Cristina; Moret, Adriane Lima Mortari

2017-12-26

The frequency modulation system is a device that helps to improve speech perception in noise and is considered the most beneficial approach to improve speech recognition in noise in cochlear implant users. According to guidelines, there is a need to perform a check before fitting the frequency modulation system. Although there are recommendations regarding the behavioral tests that should be performed at the fitting of the frequency modulation system to cochlear implant users, there are no published recommendations regarding the electroacoustic test that should be performed. Perform and determine the validity of an electroacoustic verification test for frequency modulation systems coupled to different cochlear implant speech processors. The sample included 40 participants between 5 and 18 year's users of four different models of speech processors. For the electroacoustic evaluation, we used the Audioscan Verifit device with the HA-1 coupler and the listening check devices corresponding to each speech processor model. In cases where the transparency was not achieved, a modification was made in the frequency modulation gain adjustment and we used the Brazilian version of the "Phrases in Noise Test" to evaluate the speech perception in competitive noise. It was observed that there was transparency between the frequency modulation system and the cochlear implant in 85% of the participants evaluated. After adjusting the gain of the frequency modulation receiver in the other participants, the devices showed transparency when the electroacoustic verification test was repeated. It was also observed that patients demonstrated better performance in speech perception in noise after a new adjustment, that is, in these cases; the electroacoustic transparency caused behavioral transparency. The electroacoustic evaluation protocol suggested was effective in evaluation of transparency between the frequency modulation system and the cochlear implant. Performing the adjustment of the speech processor and the frequency modulation system gain are essential when fitting this device. Copyright © 2017 Associação Brasileira de Otorrinolaringologia e Cirurgia Cérvico-Facial. Published by Elsevier Editora Ltda. All rights reserved.

Comparison of speech perception performance between Sprint/Esprit 3G and Freedom processors in children implanted with nucleus cochlear implants.

PubMed

Santarelli, Rosamaria; Magnavita, Vincenzo; De Filippi, Roberta; Ventura, Laura; Genovese, Elisabetta; Arslan, Edoardo

2009-04-01

To compare speech perception performance in children fitted with previous generation Nucleus sound processor, Sprint or Esprit 3G, and the Freedom, the most recently released system from the Cochlear Corporation that features a larger input dynamic range. Prospective intrasubject comparative study. University Medical Center. Seventeen prelingually deafened children who had received the Nucleus 24 cochlear implant and used the Sprint or Esprit 3G sound processor. Cochlear implantation with Cochlear device. Speech perception was evaluated at baseline (Sprint, n = 11; Esprit 3G, n = 6) and after 1 month's experience with the Freedom sound processor. Identification and recognition of disyllabic words and identification of vowels were performed via recorded voice in quiet (70 dB [A]), in the presence of background noise at various levels of signal-to-noise ratio (+10, +5, 0, -5) and at a soft presentation level (60 dB [A]). Consonant identification and recognition of disyllabic words, trisyllabic words, and sentences were evaluated in live voice. Frequency discrimination was measured in a subset of subjects (n = 5) by using an adaptive, 3-interval, 3-alternative, forced-choice procedure. Identification of disyllabic words administered at a soft presentation level showed a significant increase when switching to the Freedom compared with the previously worn processor in children using the Sprint or Esprit 3G. Identification and recognition of disyllabic words in the presence of background noise as well as consonant identification and sentence recognition increased significantly for the Freedom compared with the previously worn device only in children fitted with the Sprint. Frequency discrimination was significantly better when switching to the Freedom compared with the previously worn processor. Serial comparisons revealed that that speech perception performance evaluated in children aged 5 to 15 years was superior with the Freedom than previous generations of Nucleus sound processors. These differences are deemed to ensue from an increased input dynamic range, a feature that offers potentially enhanced phonemic discrimination.

[Technical advancements in cochlear implants : State of the art].

PubMed

Büchner, A; Gärtner, L

2017-04-01

Twenty years ago, cochlear implants (CI) were indicated only in cases of profound hearing loss or complete deafness. While from today's perspective the technology was clumsy and provided patients with only limited speech comprehension in quiet scenarios, successive advances in CI technology and the consequent substantial hearing improvements over time have since then resulted in continuous relaxation of indication criteria toward residual hearing. While achievements in implant and processor electronics have been one key factor for the ever-improving hearing performance, development of electro-acoustic CI systems-together with atraumatic implantation concepts-has led to enormous improvements in patients with low-frequency residual hearing. Manufactures have designed special processors with integrated hearing aid components for this patient group, which are capable of conveying acoustic and electric stimulation. A further milestone in improvement of hearing in challenging listening environments was the adoption of signal enhancement algorithms and assistive listening devices from the hearing aid industry. This article gives an overview of the current state of the art in the abovementioned areas of CI technology.

Application of a VLSI vector quantization processor to real-time speech coding

NASA Technical Reports Server (NTRS)

Davidson, G.; Gersho, A.

1986-01-01

Attention is given to a working vector quantization processor for speech coding that is based on a first-generation VLSI chip which efficiently performs the pattern-matching operation needed for the codebook search process (CPS). Using this chip, the CPS architecture has been successfully incorporated into a compact, single-board Vector PCM implementation operating at 7-18 kbits/sec. A real time Adaptive Vector Predictive Coder system using the CPS has also been implemented.

A Deep Denoising Autoencoder Approach to Improving the Intelligibility of Vocoded Speech in Cochlear Implant Simulation.

PubMed

Lai, Ying-Hui; Chen, Fei; Wang, Syu-Siang; Lu, Xugang; Tsao, Yu; Lee, Chin-Hui

2017-07-01

In a cochlear implant (CI) speech processor, noise reduction (NR) is a critical component for enabling CI users to attain improved speech perception under noisy conditions. Identifying an effective NR approach has long been a key topic in CI research. Recently, a deep denoising autoencoder (DDAE) based NR approach was proposed and shown to be effective in restoring clean speech from noisy observations. It was also shown that DDAE could provide better performance than several existing NR methods in standardized objective evaluations. Following this success with normal speech, this paper further investigated the performance of DDAE-based NR to improve the intelligibility of envelope-based vocoded speech, which simulates speech signal processing in existing CI devices. We compared the performance of speech intelligibility between DDAE-based NR and conventional single-microphone NR approaches using the noise vocoder simulation. The results of both objective evaluations and listening test showed that, under the conditions of nonstationary noise distortion, DDAE-based NR yielded higher intelligibility scores than conventional NR approaches. This study confirmed that DDAE-based NR could potentially be integrated into a CI processor to provide more benefits to CI users under noisy conditions.

Designing a Humane Multimedia Interface for the Visually Impaired.

ERIC Educational Resources Information Center

Ghaoui, Claude; Mann, M.; Ng, Eng Huat

2001-01-01

Promotes the provision of interfaces that allow users to access most of the functionality of existing graphical user interfaces (GUI) using speech. Uses the design of a speech control tool that incorporates speech recognition and synthesis into existing packaged software such as Teletext, the Internet, or a word processor. (Contains 22…

Gender and speaker identification as a function of the number of channels in spectrally reduced speech

NASA Astrophysics Data System (ADS)

Gonzalez, Julio; Oliver, Juan C.

2005-07-01

Considerable research on speech intelligibility for cochlear-implant users has been conducted using acoustic simulations with normal-hearing subjects. However, some relevant topics about perception through cochlear implants remain scantly explored. The present study examined the perception by normal-hearing subjects of gender and identity of a talker as a function of the number of channels in spectrally reduced speech. Two simulation strategies were compared. They were implemented by two different processors that presented signals as either the sum of sine waves at the center of the channels or as the sum of noise bands. In Experiment 1, 15 subjects determined the gender of 40 talkers (20 males + 20 females) from a natural utterance processed through 3, 4, 5, 6, 8, 10, 12, and 16 channels with both processors. In Experiment 2, 56 subjects matched a natural sentence uttered by 10 talkers with the corresponding simulation replicas processed through 3, 4, 8, and 16 channels for each processor. In Experiment 3, 72 subjects performed the same task but different sentences were used for natural and processed stimuli. A control Experiment 4 was conducted to equate the processing steps between the two simulation strategies. Results showed that gender and talker identification was better for the sine-wave processor, and that performance through the noise-band processor was more sensitive to the number of channels. Implications and possible explanations for the superiority of sine-wave simulations are discussed.

Evaluation of speech reception threshold in noise in young Cochlear™ Nucleus® system 6 implant recipients using two different digital remote microphone technologies and a speech enhancement sound processing algorithm.

PubMed

Razza, Sergio; Zaccone, Monica; Meli, Aannalisa; Cristofari, Eliana

2017-12-01

Children affected by hearing loss can experience difficulties in challenging and noisy environments even when deafness is corrected by Cochlear implant (CI) devices. These patients have a selective attention deficit in multiple listening conditions. At present, the most effective ways to improve the performance of speech recognition in noise consists of providing CI processors with noise reduction algorithms and of providing patients with bilateral CIs. The aim of this study was to compare speech performances in noise, across increasing noise levels, in CI recipients using two kinds of wireless remote-microphone radio systems that use digital radio frequency transmission: the Roger Inspiro accessory and the Cochlear Wireless Mini Microphone accessory. Eleven Nucleus Cochlear CP910 CI young user subjects were studied. The signal/noise ratio, at a speech reception threshold (SRT) value of 50%, was measured in different conditions for each patient: with CI only, with the Roger or with the MiniMic accessory. The effect of the application of the SNR-noise reduction algorithm in each of these conditions was also assessed. The tests were performed with the subject positioned in front of the main speaker, at a distance of 2.5 m. Another two speakers were positioned at 3.50 m. The main speaker at 65 dB issued disyllabic words. Babble noise signal was delivered through the other speakers, with variable intensity. The use of both wireless remote microphones improved the SRT results. Both systems improved gain of speech performances. The gain was higher with the Mini Mic system (SRT = -4.76) than the Roger system (SRT = -3.01). The addition of the NR algorithm did not statistically further improve the results. There is significant improvement in speech recognition results with both wireless digital remote microphone accessories, in particular with the Mini Mic system when used with the CP910 processor. The use of a remote microphone accessory surpasses the benefit of application of NR algorithm. Copyright © 2017. Published by Elsevier B.V.

An investigation of potential applications of OP-SAPS: Operational Sampled Analog Processors

NASA Technical Reports Server (NTRS)

Parrish, E. A.; Mcvey, E. S.

1977-01-01

The application of OP-SAP's (operational sampled analog processors) in pattern recognition system is summarized. Areas investigated include: (1) human face recognition; (2) a high-speed programmable transversal filter system; (3) discrete word (speech) recognition; and (4) a resolution enhancement system.

Everyday listening questionnaire: correlation between subjective hearing and objective performance.

PubMed

Brendel, Martina; Frohne-Buechner, Carolin; Lesinski-Schiedat, Anke; Lenarz, Thomas; Buechner, Andreas

2014-01-01

Clinical experience has demonstrated that speech understanding by cochlear implant (CI) recipients has improved over recent years with the development of new technology. The Everyday Listening Questionnaire 2 (ELQ 2) was designed to collect information regarding the challenges faced by CI recipients in everyday listening. The aim of this study was to compare self-assessment of CI users using ELQ 2 with objective speech recognition measures and to compare results between users of older and newer coding strategies. During their regular clinical review appointments a group of representative adult CI recipients implanted with the Advanced Bionics implant system were asked to complete the questionnaire. The first 100 patients who agreed to participate in this survey were recruited independent of processor generation and speech coding strategy. Correlations between subjectively scored hearing performance in everyday listening situations and objectively measured speech perception abilities were examined relative to the speech coding strategies used. When subjects were grouped by strategy there were significant differences between users of older 'standard' strategies and users of the newer, currently available strategies (HiRes and HiRes 120), especially in the categories of telephone use and music perception. Significant correlations were found between certain subjective ratings and the objective speech perception data in noise. There is a good correlation between subjective and objective data. Users of more recent speech coding strategies tend to have fewer problems in difficult hearing situations.

The ability of cochlear implant users to use temporal envelope cues recovered from speech frequency modulation.

PubMed

Won, Jong Ho; Lorenzi, Christian; Nie, Kaibao; Li, Xing; Jameyson, Elyse M; Drennan, Ward R; Rubinstein, Jay T

2012-08-01

Previous studies have demonstrated that normal-hearing listeners can understand speech using the recovered "temporal envelopes," i.e., amplitude modulation (AM) cues from frequency modulation (FM). This study evaluated this mechanism in cochlear implant (CI) users for consonant identification. Stimuli containing only FM cues were created using 1, 2, 4, and 8-band FM-vocoders to determine if consonant identification performance would improve as the recovered AM cues become more available. A consistent improvement was observed as the band number decreased from 8 to 1, supporting the hypothesis that (1) the CI sound processor generates recovered AM cues from broadband FM, and (2) CI users can use the recovered AM cues to recognize speech. The correlation between the intact and the recovered AM components at the output of the sound processor was also generally higher when the band number was low, supporting the consonant identification results. Moreover, CI subjects who were better at using recovered AM cues from broadband FM cues showed better identification performance with intact (unprocessed) speech stimuli. This suggests that speech perception performance variability in CI users may be partly caused by differences in their ability to use AM cues recovered from FM speech cues.

Speech production in experienced cochlear implant users undergoing short-term auditory deprivation

NASA Astrophysics Data System (ADS)

Greenman, Geoffrey; Tjaden, Kris; Kozak, Alexa T.

2005-09-01

This study examined the effect of short-term auditory deprivation on the speech production of five postlingually deafened women, all of whom were experienced cochlear implant users. Each cochlear implant user, as well as age and gender matched control speakers, produced CVC target words embedded in a reading passage. Speech samples for the deafened adults were collected on two separate occasions. First, the speakers were recorded after wearing their speech processor consistently for at least two to three hours prior to recording (implant ``ON''). The second recording occurred when the speakers had their speech processors turned off for approximately ten to twelve hours prior to recording (implant ``OFF''). Acoustic measures, including fundamental frequency (F0), the first (F1) and second (F2) formants of the vowels, vowel space area, vowel duration, spectral moments of the consonants, as well as utterance duration and sound pressure level (SPL) across the entire utterance were analyzed in both speaking conditions. For each implant speaker, acoustic measures will be compared across implant ``ON'' and implant ``OFF'' speaking conditions, and will also be compared to data obtained from normal hearing speakers.

Assessment of directionality performances: comparison between Freedom and CP810 sound processors.

PubMed

Razza, Sergio; Albanese, Greta; Ermoli, Lucilla; Zaccone, Monica; Cristofari, Eliana

2013-10-01

To compare speech recognition in noise for the Nucleus Freedom and CP810 sound processors using different directional settings among those available in the SmartSound portfolio. Single-subject, repeated measures study. Tertiary care referral center. Thirty-one monoaurally and binaurally implanted subjects (24 children and 7 adults) were enrolled. They were all experienced Nucleus Freedom sound processor users and achieved a 100% open set word recognition score in quiet listening conditions. Each patient was fitted with the Freedom and the CP810 processor. The program setting incorporated Adaptive Dynamic Range Optimization (ADRO) and adopted the directional algorithm BEAM (both devices) and ZOOM (only on CP810). Speech reception threshold (SRT) was assessed in a free-field layout, with disyllabic word list and interfering multilevel babble noise in the 3 different pre-processing configurations. On average, CP810 improved significantly patients' SRTs as compared to Freedom SP after 1 hour of use. Instead, no significant difference was observed in patients' SRT between the BEAM and the ZOOM algorithm fitted in the CP810 processor. The results suggest that hardware developments achieved in the design of CP810 allow an immediate and relevant directional advantage as compared to the previous-generation Freedom device.

Using Temporal Modulation Sensitivity to Select Stimulation Sites for Processor MAPs in Cochlear Implant Listeners

PubMed Central

Garadat, Soha N.; Zwolan, Teresa A.; Pfingst, Bryan E.

2013-01-01

Previous studies in our laboratory showed that temporal acuity as assessed by modulation detection thresholds (MDTs) varied across activation sites and that this site-to-site variability was subject specific. Using two 10-channel MAPs, the previous experiments showed that processor MAPs that had better across-site mean (ASM) MDTs yielded better speech recognition than MAPs with poorer ASM MDTs tested in the same subject. The current study extends our earlier work on developing more optimal fitting strategies to test the feasibility of using a site-selection approach in the clinical domain. This study examined the hypothesis that revising the clinical speech processor MAP for cochlear implant (CI) recipients by turning off selected sites that have poorer temporal acuity and reallocating frequencies to the remaining electrodes would lead to improved speech recognition. Twelve CI recipients participated in the experiments. We found that site selection procedure based on MDTs in the presence of a masker resulted in improved performance on consonant recognition and recognition of sentences in noise. In contrast, vowel recognition was poorer with the experimental MAP than with the clinical MAP, possibly due to reduced spectral resolution when sites were removed from the experimental MAP. Overall, these results suggest a promising path for improving recipient outcomes using personalized processor-fitting strategies based on a psychophysical measure of temporal acuity. PMID:23881208

Speech recognition for embedded automatic positioner for laparoscope

NASA Astrophysics Data System (ADS)

Chen, Xiaodong; Yin, Qingyun; Wang, Yi; Yu, Daoyin

2014-07-01

In this paper a novel speech recognition methodology based on Hidden Markov Model (HMM) is proposed for embedded Automatic Positioner for Laparoscope (APL), which includes a fixed point ARM processor as the core. The APL system is designed to assist the doctor in laparoscopic surgery, by implementing the specific doctor's vocal control to the laparoscope. Real-time respond to the voice commands asks for more efficient speech recognition algorithm for the APL. In order to reduce computation cost without significant loss in recognition accuracy, both arithmetic and algorithmic optimizations are applied in the method presented. First, depending on arithmetic optimizations most, a fixed point frontend for speech feature analysis is built according to the ARM processor's character. Then the fast likelihood computation algorithm is used to reduce computational complexity of the HMM-based recognition algorithm. The experimental results show that, the method shortens the recognition time within 0.5s, while the accuracy higher than 99%, demonstrating its ability to achieve real-time vocal control to the APL.

Interdependence of linguistic and indexical speech perception skills in school-age children with early cochlear implantation.

PubMed

Geers, Ann E; Davidson, Lisa S; Uchanski, Rosalie M; Nicholas, Johanna G

2013-09-01

This study documented the ability of experienced pediatric cochlear implant (CI) users to perceive linguistic properties (what is said) and indexical attributes (emotional intent and talker identity) of speech, and examined the extent to which linguistic (LSP) and indexical (ISP) perception skills are related. Preimplant-aided hearing, age at implantation, speech processor technology, CI-aided thresholds, sequential bilateral cochlear implantation, and academic integration with hearing age-mates were examined for their possible relationships to both LSP and ISP skills. Sixty 9- to 12-year olds, first implanted at an early age (12 to 38 months), participated in a comprehensive test battery that included the following LSP skills: (1) recognition of monosyllabic words at loud and soft levels, (2) repetition of phonemes and suprasegmental features from nonwords, and (3) recognition of key words from sentences presented within a noise background, and the following ISP skills: (1) discrimination of across-gender and within-gender (female) talkers and (2) identification and discrimination of emotional content from spoken sentences. A group of 30 age-matched children without hearing loss completed the nonword repetition, and talker- and emotion-perception tasks for comparison. Word-recognition scores decreased with signal level from a mean of 77% correct at 70 dB SPL to 52% at 50 dB SPL. On average, CI users recognized 50% of key words presented in sentences that were 9.8 dB above background noise. Phonetic properties were repeated from nonword stimuli at about the same level of accuracy as suprasegmental attributes (70 and 75%, respectively). The majority of CI users identified emotional content and differentiated talkers significantly above chance levels. Scores on LSP and ISP measures were combined into separate principal component scores and these components were highly correlated (r = 0.76). Both LSP and ISP component scores were higher for children who received a CI at the youngest ages, upgraded to more recent CI technology and had lower CI-aided thresholds. Higher scores, for both LSP and ISP components, were also associated with higher language levels and mainstreaming at younger ages. Higher ISP scores were associated with better social skills. Results strongly support a link between indexical and linguistic properties in perceptual analysis of speech. These two channels of information appear to be processed together in parallel by the auditory system and are inseparable in perception. Better speech performance, for both linguistic and indexical perception, is associated with younger age at implantation and use of more recent speech processor technology. Children with better speech perception demonstrated better spoken language, earlier academic mainstreaming, and placement in more typically sized classrooms (i.e., >20 students). Well-developed social skills were more highly associated with the ability to discriminate the nuances of talker identity and emotion than with the ability to recognize words and sentences through listening. The extent to which early cochlear implantation enabled these early-implanted children to make use of both linguistic and indexical properties of speech influenced not only their development of spoken language, but also their ability to function successfully in a hearing world.

Interdependence of Linguistic and Indexical Speech Perception Skills in School-Aged Children with Early Cochlear Implantation

PubMed Central

Geers, Ann; Davidson, Lisa; Uchanski, Rosalie; Nicholas, Johanna

2013-01-01

Objectives This study documented the ability of experienced pediatric cochlear implant (CI) users to perceive linguistic properties (what is said) and indexical attributes (emotional intent and talker identity) of speech, and examined the extent to which linguistic (LSP) and indexical (ISP) perception skills are related. Pre-implant aided hearing, age at implantation, speech processor technology, CI-aided thresholds, sequential bilateral cochlear implantation, and academic integration with hearing age-mates were examined for their possible relationships to both LSP and ISP skills. Design Sixty 9–12 year olds, first implanted at an early age (12–38 months), participated in a comprehensive test battery that included the following LSP skills: 1) recognition of monosyllabic words at loud and soft levels, 2) repetition of phonemes and suprasegmental features from non-words, and 3) recognition of keywords from sentences presented within a noise background, and the following ISP skills: 1) discrimination of male from female and female from female talkers and 2) identification and discrimination of emotional content from spoken sentences. A group of 30 age-matched children without hearing loss completed the non-word repetition, and talker- and emotion-perception tasks for comparison. Results Word recognition scores decreased with signal level from a mean of 77% correct at 70 dB SPL to 52% at 50 dB SPL. On average, CI users recognized 50% of keywords presented in sentences that were 9.8 dB above background noise. Phonetic properties were repeated from non-word stimuli at about the same level of accuracy as suprasegmental attributes (70% and 75%, respectively). The majority of CI users identified emotional content and differentiated talkers significantly above chance levels. Scores on LSP and ISP measures were combined into separate principal component scores and these components were highly correlated (r = .76). Both LSP and ISP component scores were higher for children who received a CI at the youngest ages, upgraded to more recent CI technology and had lower CI-aided thresholds. Higher scores, for both LSP and ISP components, were also associated with higher language levels and mainstreaming at younger ages. Higher ISP scores were associated with better social skills. Conclusions Results strongly support a link between indexical and linguistic properties in perceptual analysis of speech. These two channels of information appear to be processed together in parallel by the auditory system and are inseparable in perception. Better speech performance, for both linguistic and indexical perception, is associated with younger age at implantation and use of more recent speech processor technology. Children with better speech perception demonstrated better spoken language, earlier academic mainstreaming, and placement in more typically-sized classrooms (i.e., >20 students). Well-developed social skills were more highly associated with the ability to discriminate the nuances of talker identity and emotion than with the ability to recognize words and sentences through listening. The extent to which early cochlear implantation enabled these early-implanted children to make use of both linguistic and indexical properties of speech influenced not only their development of spoken language, but also their ability to function successfully in a hearing world. PMID:23652814

Internship Abstract and Final Reflection

NASA Technical Reports Server (NTRS)

Sandor, Edward

2016-01-01

The primary objective for this internship is the evaluation of an embedded natural language processor (NLP) as a way to introduce voice control into future space suits. An embedded natural language processor would provide an astronaut hands-free control for making adjustments to the environment of the space suit and checking status of consumables procedures and navigation. Additionally, the use of an embedded NLP could potentially reduce crew fatigue, increase the crewmember's situational awareness during extravehicular activity (EVA) and improve the ability to focus on mission critical details. The use of an embedded NLP may be valuable for other human spaceflight applications desiring hands-free control as well. An embedded NLP is unique because it is a small device that performs language tasks, including speech recognition, which normally require powerful processors. The dedicated device could perform speech recognition locally with a smaller form-factor and lower power consumption than traditional methods.

Effects of Talker Variability on Vowel Recognition in Cochlear Implants

ERIC Educational Resources Information Center

Chang, Yi-ping; Fu, Qian-Jie

2006-01-01

Purpose: To investigate the effects of talker variability on vowel recognition by cochlear implant (CI) users and by normal-hearing (NH) participants listening to 4-channel acoustic CI simulations. Method: CI users were tested with their clinically assigned speech processors. For NH participants, 3 CI processors were simulated, using different…

Across-site patterns of modulation detection: Relation to speech recognitiona)

PubMed Central

Garadat, Soha N.; Zwolan, Teresa A.; Pfingst, Bryan E.

2012-01-01

The aim of this study was to identify across-site patterns of modulation detection thresholds (MDTs) in subjects with cochlear implants and to determine if removal of sites with the poorest MDTs from speech processor programs would result in improved speech recognition. Five hundred millisecond trains of symmetric-biphasic pulses were modulated sinusoidally at 10 Hz and presented at a rate of 900 pps using monopolar stimulation. Subjects were asked to discriminate a modulated pulse train from an unmodulated pulse train for all electrodes in quiet and in the presence of an interleaved unmodulated masker presented on the adjacent site. Across-site patterns of masked MDTs were then used to construct two 10-channel MAPs such that one MAP consisted of sites with the best masked MDTs and the other MAP consisted of sites with the worst masked MDTs. Subjects’ speech recognition skills were compared when they used these two different MAPs. Results showed that MDTs were variable across sites and were elevated in the presence of a masker by various amounts across sites. Better speech recognition was observed when the processor MAP consisted of sites with best masked MDTs, suggesting that temporal modulation sensitivity has important contributions to speech recognition with a cochlear implant. PMID:22559376

Telephone speech comprehension in children with multichannel cochlear implants.

PubMed

Aronson, L; Estienne, P; Arauz, S L; Pallante, S A

1997-11-01

Telephone speech comprehension is being evaluated in six prelingually deaf children implanted with the Nucleus 22 prosthesis fitted with the Speak strategy. All of them have had at least 1.5 years of experience with their implant. When the tests began, they had already had at least 2 months' experience with the same map in their speech processor. The children were trained in the use of the telephone as part of the rehabilitation program. None of them used it regularly but as a game that they found very entertaining. A special battery, the Bate-fon (batería para teléfono = telephone battery), was designed for training and evaluation purposes. It includes the five Spanish vowels in isolation, diphthongs, onomatopoetic animal voices, two-syllable, and three-syllable words. The tests were administered 1.5-2 years after the switch-on of their speech processor. Standard acoustic telephone coupling was used. The speech material was presented to the child on colored cards. Stimuli were presented twice. Children were informed when the response was incorrect. Averaged results indicated that the percentages of correct responses for all the speech material increase in the second presentation. All children have shown some degree of telephone communication abilities. As a result of the training, some of the children are using the telephone to communicate with their families.

The ability of cochlear implant users to use temporal envelope cues recovered from speech frequency modulationa

PubMed Central

Won, Jong Ho; Lorenzi, Christian; Nie, Kaibao; Li, Xing; Jameyson, Elyse M.; Drennan, Ward R.; Rubinstein, Jay T.

2012-01-01

Previous studies have demonstrated that normal-hearing listeners can understand speech using the recovered “temporal envelopes,” i.e., amplitude modulation (AM) cues from frequency modulation (FM). This study evaluated this mechanism in cochlear implant (CI) users for consonant identification. Stimuli containing only FM cues were created using 1, 2, 4, and 8-band FM-vocoders to determine if consonant identification performance would improve as the recovered AM cues become more available. A consistent improvement was observed as the band number decreased from 8 to 1, supporting the hypothesis that (1) the CI sound processor generates recovered AM cues from broadband FM, and (2) CI users can use the recovered AM cues to recognize speech. The correlation between the intact and the recovered AM components at the output of the sound processor was also generally higher when the band number was low, supporting the consonant identification results. Moreover, CI subjects who were better at using recovered AM cues from broadband FM cues showed better identification performance with intact (unprocessed) speech stimuli. This suggests that speech perception performance variability in CI users may be partly caused by differences in their ability to use AM cues recovered from FM speech cues. PMID:22894230

The effect of instantaneous input dynamic range setting on the speech perception of children with the nucleus 24 implant.

PubMed

Davidson, Lisa S; Skinner, Margaret W; Holstad, Beth A; Fears, Beverly T; Richter, Marie K; Matusofsky, Margaret; Brenner, Christine; Holden, Timothy; Birath, Amy; Kettel, Jerrica L; Scollie, Susan

2009-06-01

The purpose of this study was to examine the effects of a wider instantaneous input dynamic range (IIDR) setting on speech perception and comfort in quiet and noise for children wearing the Nucleus 24 implant system and the Freedom speech processor. In addition, children's ability to understand soft and conversational level speech in relation to aided sound-field thresholds was examined. Thirty children (age, 7 to 17 years) with the Nucleus 24 cochlear implant system and the Freedom speech processor with two different IIDR settings (30 versus 40 dB) were tested on the Consonant Nucleus Consonant (CNC) word test at 50 and 60 dB SPL, the Bamford-Kowal-Bench Speech in Noise Test, and a loudness rating task for four-talker speech noise. Aided thresholds for frequency-modulated tones, narrowband noise, and recorded Ling sounds were obtained with the two IIDRs and examined in relation to CNC scores at 50 dB SPL. Speech Intelligibility Indices were calculated using the long-term average speech spectrum of the CNC words at 50 dB SPL measured at each test site and aided thresholds. Group mean CNC scores at 50 dB SPL with the 40 IIDR were significantly higher (p < 0.001) than with the 30 IIDR. Group mean CNC scores at 60 dB SPL, loudness ratings, and the signal to noise ratios-50 for Bamford-Kowal-Bench Speech in Noise Test were not significantly different for the two IIDRs. Significantly improved aided thresholds at 250 to 6000 Hz as well as higher Speech Intelligibility Indices afforded improved audibility for speech presented at soft levels (50 dB SPL). These results indicate that an increased IIDR provides improved word recognition for soft levels of speech without compromising comfort of higher levels of speech sounds or sentence recognition in noise.

Wireless and acoustic hearing with bone-anchored hearing devices.

PubMed

Bosman, Arjan J; Mylanus, Emmanuel A M; Hol, Myrthe K S; Snik, Ad F M

2015-07-01

The efficacy of wireless connectivity in bone-anchored hearing was studied by comparing the wireless and acoustic performance of the Ponto Plus sound processor from Oticon Medical relative to the acoustic performance of its predecessor, the Ponto Pro. Nineteen subjects with more than two years' experience with a bone-anchored hearing device were included. Thirteen subjects were fitted unilaterally and six bilaterally. Subjects served as their own control. First, subjects were tested with the Ponto Pro processor. After a four-week acclimatization period performance the Ponto Plus processor was measured. In the laboratory wireless and acoustic input levels were made equal. In daily life equal settings of wireless and acoustic input were used when watching TV, however when using the telephone the acoustic input was reduced by 9 dB relative to the wireless input. Speech scores for microphone with Ponto Pro and for both input modes of the Ponto Plus processor were essentially equal when equal input levels of wireless and microphone inputs were used. Only the TV-condition showed a statistically significant (p <5%) lower speech reception threshold for wireless relative to microphone input. In real life, evaluation of speech quality, speech intelligibility in quiet and noise, and annoyance by ambient noise, when using landline phone, mobile telephone, and watching TV showed a clear preference (p <1%) for the Ponto Plus system with streamer over the microphone input. Due to the small number of respondents with landline phone (N = 7) the result for noise annoyance was only significant at the 5% level. Equal input levels for acoustic and wireless inputs results in equal speech scores, showing a (near) equivalence for acoustic and wireless sound transmission with Ponto Pro and Ponto Plus. The default 9-dB difference between microphone and wireless input when using the telephone results in a substantial wireless benefit when using the telephone. The preference of wirelessly transmitted audio when watching TV can be attributed to the relatively poor sound quality of backward facing loudspeakers in flat screen TVs. The ratio of wireless and acoustic input can be easily set to the user's preference with the streamer's volume control.

Speech perception comparisons using an implanted and an external microphone in existing cochlear implant users.

PubMed

Jenkins, Herman A; Uhler, Kristin

2012-01-01

To compare the speech understanding abilities of cochlear implant listeners using 2 microphone technologies, the Otologics fully implantable Carina and the Cochlear Freedom microphones. Feasibility study using direct comparison of the 2 microphones, nonrandomized and nonblinded within case studies. Tertiary referral center hospital outpatient clinic. Four subjects with greater than 1 year of unilateral listening experience with the Freedom Cochlear Implant and a CNC word score higher than 40%. A Carina microphone coupled to a percutaneous plug was implanted on the ipsilateral side of the cochlear implant. Two months were allowed for healing before connecting to the Carina microphone. The percutaneous plug was connected to a body worn external processor with output leads inserted into the auxiliary port of the Freedom processor. Subjects were instructed to use each of the 2 microphones for half of their daily implant use. Aided pure tone thresholds, consonant-nucleus-consonant (CNC), Bamford-Kowel-Bench Speech in Noise test (BKN-SIN), and Abbreviated Profile of Hearing Aid Benefit. All subjects had sound perceptions using both microphones. The loudness and quality of the sound was judged to be poorer with the Carina in the first 2 subjects. The latter 2 demonstrated essential equivalence in the second two listeners, with the exception of the Abbreviated Profile of Hearing Aid Benefit reporting greater percentage of problems for the Carina in the background noise situation for subject 0011-003PP. CNC word scores were better with the Freedom than the Carina in all 4 subjects. The latter 2 showed improved speech perception abilities with the Carina, compared with the first 2. The BKB-SIN showed consistently better results with the Freedom in noise. Early observations indicate that it is potentially feasible to use the fully implanted Carina microphone with the Freedom Cochlear Implant. The authors would anticipate that outcomes would improve as more knowledge is gained in signal processing and with the fabrication of an integrated device.

On the Perception of Speech Sounds as Biologically Significant Signals1,2

PubMed Central

Pisoni, David B.

2012-01-01

This paper reviews some of the major evidence and arguments currently available to support the view that human speech perception may require the use of specialized neural mechanisms for perceptual analysis. Experiments using synthetically produced speech signals with adults are briefly summarized and extensions of these results to infants and other organisms are reviewed with an emphasis towards detailing those aspects of speech perception that may require some need for specialized species-specific processors. Finally, some comments on the role of early experience in perceptual development are provided as an attempt to identify promising areas of new research in speech perception. PMID:399200

Is complex signal processing for bone conduction hearing aids useful?

PubMed

Kompis, Martin; Kurz, Anja; Pfiffner, Flurin; Senn, Pascal; Arnold, Andreas; Caversaccio, Marco

2014-05-01

To establish whether complex signal processing is beneficial for users of bone anchored hearing aids. Review and analysis of two studies from our own group, each comparing a speech processor with basic digital signal processing (either Baha Divino or Baha Intenso) and a processor with complex digital signal processing (either Baha BP100 or Baha BP110 power). The main differences between basic and complex signal processing are the number of audiologist accessible frequency channels and the availability and complexity of the directional multi-microphone noise reduction and loudness compression systems. Both studies show a small, statistically non-significant improvement of speech understanding in quiet with the complex digital signal processing. The average improvement for speech in noise is +0.9 dB, if speech and noise are emitted both from the front of the listener. If noise is emitted from the rear and speech from the front of the listener, the advantage of the devices with complex digital signal processing as opposed to those with basic signal processing increases, on average, to +3.2 dB (range +2.3 … +5.1 dB, p ≤ 0.0032). Complex digital signal processing does indeed improve speech understanding, especially in noise coming from the rear. This finding has been supported by another study, which has been published recently by a different research group. When compared to basic digital signal processing, complex digital signal processing can increase speech understanding of users of bone anchored hearing aids. The benefit is most significant for speech understanding in noise.

Binaural unmasking of multi-channel stimuli in bilateral cochlear implant users.

PubMed

Van Deun, Lieselot; van Wieringen, Astrid; Francart, Tom; Büchner, Andreas; Lenarz, Thomas; Wouters, Jan

2011-10-01

Previous work suggests that bilateral cochlear implant users are sensitive to interaural cues if experimental speech processors are used to preserve accurate interaural information in the electrical stimulation pattern. Binaural unmasking occurs in adults and children when an interaural delay is applied to the envelope of a high-rate pulse train. Nevertheless, for speech perception, binaural unmasking benefits have not been demonstrated consistently, even with coordinated stimulation at both ears. The present study aimed at bridging the gap between basic psychophysical performance on binaural signal detection tasks on the one hand and binaural perception of speech in noise on the other hand. Therefore, binaural signal detection was expanded to multi-channel stimulation and biologically relevant interaural delays. A harmonic complex, consisting of three sinusoids (125, 250, and 375 Hz), was added to three 125-Hz-wide noise bands centered on the sinusoids. When an interaural delay of 700 μs was introduced, an average BMLD of 3 dB was established. Outcomes are promising in view of real-life benefits. Future research should investigate the generalization of the observed benefits for signal detection to speech perception in everyday listening situations and determine the importance of coordination of bilateral speech processors and accentuation of envelope cues.

Amplitude modulation detection with concurrent frequency modulation.

PubMed

Nagaraj, Naveen K

2016-09-01

Human speech consists of concomitant temporal modulations in amplitude and frequency that are crucial for speech perception. In this study, amplitude modulation (AM) detection thresholds were measured for 550 and 5000 Hz carriers with and without concurrent frequency modulation (FM), at AM rates crucial for speech perception. Results indicate that adding 40 Hz FM interferes with AM detection, more so for 5000 Hz carrier and for frequency deviations exceeding the critical bandwidth of the carrier frequency. These findings suggest that future cochlear implant processors, encoding speech fine-structures may consider limiting the FM to narrow bandwidth and to low frequencies.

A 4.8 kbps code-excited linear predictive coder

NASA Technical Reports Server (NTRS)

Tremain, Thomas E.; Campbell, Joseph P., Jr.; Welch, Vanoy C.

1988-01-01

A secure voice system STU-3 capable of providing end-to-end secure voice communications (1984) was developed. The terminal for the new system will be built around the standard LPC-10 voice processor algorithm. The performance of the present STU-3 processor is considered to be good, its response to nonspeech sounds such as whistles, coughs and impulse-like noises may not be completely acceptable. Speech in noisy environments also causes problems with the LPC-10 voice algorithm. In addition, there is always a demand for something better. It is hoped that LPC-10's 2.4 kbps voice performance will be complemented with a very high quality speech coder operating at a higher data rate. This new coder is one of a number of candidate algorithms being considered for an upgraded version of the STU-3 in late 1989. The problems of designing a code-excited linear predictive (CELP) coder to provide very high quality speech at a 4.8 kbps data rate that can be implemented on today's hardware are considered.

GPU-based Parallel Application Design for Emerging Mobile Devices

NASA Astrophysics Data System (ADS)

Gupta, Kshitij

A revolution is underway in the computing world that is causing a fundamental paradigm shift in device capabilities and form-factor, with a move from well-established legacy desktop/laptop computers to mobile devices in varying sizes and shapes. Amongst all the tasks these devices must support, graphics has emerged as the 'killer app' for providing a fluid user interface and high-fidelity game rendering, effectively making the graphics processor (GPU) one of the key components in (present and future) mobile systems. By utilizing the GPU as a general-purpose parallel processor, this dissertation explores the GPU computing design space from an applications standpoint, in the mobile context, by focusing on key challenges presented by these devices---limited compute, memory bandwidth, and stringent power consumption requirements---while improving the overall application efficiency of the increasingly important speech recognition workload for mobile user interaction. We broadly partition trends in GPU computing into four major categories. We analyze hardware and programming model limitations in current-generation GPUs and detail an alternate programming style called Persistent Threads, identify four use case patterns, and propose minimal modifications that would be required for extending native support. We show how by manually extracting data locality and altering the speech recognition pipeline, we are able to achieve significant savings in memory bandwidth while simultaneously reducing the compute burden on GPU-like parallel processors. As we foresee GPU computing to evolve from its current 'co-processor' model into an independent 'applications processor' that is capable of executing complex work independently, we create an alternate application framework that enables the GPU to handle all control-flow dependencies autonomously at run-time while minimizing host involvement to just issuing commands, that facilitates an efficient application implementation. Finally, as compute and communication capabilities of mobile devices improve, we analyze energy implications of processing speech recognition locally (on-chip) and offloading it to servers (in-cloud).

Fine Structure Processing improves speech perception as well as objective and subjective benefits in pediatric MED-EL COMBI 40+ users.

PubMed

Lorens, Artur; Zgoda, Małgorzata; Obrycka, Anita; Skarżynski, Henryk

2010-12-01

Presently, there are only few studies examining the benefits of fine structure information in coding strategies. Against this background, this study aims to assess the objective and subjective performance of children experienced with the C40+ cochlear implant using the CIS+ coding strategy who were upgraded to the OPUS 2 processor using FSP and HDCIS. In this prospective study, 60 children with more than 3.5 years of experience with the C40+ cochlear implant were upgraded to the OPUS 2 processor and fit and tested with HDCIS (Interval I). After 3 months of experience with HDCIS, they were fit with the FSP coding strategy (Interval II) and tested with all strategies (FSP, HDCIS, CIS+). After an additional 3-4 months, they were assessed on all three strategies and asked to choose their take-home strategy (Interval III). The children were tested using the Adaptive Auditory Speech Test which measures speech reception threshold (SRT) in quiet and noise at each test interval. The children were also asked to rate on a Visual Analogue Scale their satisfaction and coding strategy preference when listening to speech and a pop song. However, since not all tests could be performed at one single visit, some children were not able complete all tests at all intervals. At the study endpoint, speech in quiet showed a significant difference in SRT of 1.0 dB between FSP and HDCIS, with FSP performing better. FSP proved a better strategy compared with CIS+, showing lower SRT results of 5.2 dB. Speech in noise tests showed FSP to be significantly better than CIS+ by 0.7 dB, and HDCIS to be significantly better than CIS+ by 0.8 dB. Both satisfaction and coding strategy preference ratings also revealed that FSP and HDCIS strategies were better than CIS+ strategy when listening to speech and music. FSP was better than HDCIS when listening to speech. This study demonstrates that long-term pediatric users of the COMBI 40+ are able to upgrade to a newer processor and coding strategy without compromising their listening performance and even improving their performance with FSP after a short time of experience. Copyright © 2010 Elsevier Ireland Ltd. All rights reserved.

Combined electric and acoustic hearing performance with Zebra® speech processor: speech reception, place, and temporal coding evaluation.

PubMed

Vaerenberg, Bart; Péan, Vincent; Lesbros, Guillaume; De Ceulaer, Geert; Schauwers, Karen; Daemers, Kristin; Gnansia, Dan; Govaerts, Paul J

2013-06-01

To assess the auditory performance of Digisonic(®) cochlear implant users with electric stimulation (ES) and electro-acoustic stimulation (EAS) with special attention to the processing of low-frequency temporal fine structure. Six patients implanted with a Digisonic(®) SP implant and showing low-frequency residual hearing were fitted with the Zebra(®) speech processor providing both electric and acoustic stimulation. Assessment consisted of monosyllabic speech identification tests in quiet and in noise at different presentation levels, and a pitch discrimination task using harmonic and disharmonic intonating complex sounds ( Vaerenberg et al., 2011 ). These tests investigate place and time coding through pitch discrimination. All tasks were performed with ES only and with EAS. Speech results in noise showed significant improvement with EAS when compared to ES. Whereas EAS did not yield better results in the harmonic intonation test, the improvements in the disharmonic intonation test were remarkable, suggesting better coding of pitch cues requiring phase locking. These results suggest that patients with residual hearing in the low-frequency range still have good phase-locking capacities, allowing them to process fine temporal information. ES relies mainly on place coding but provides poor low-frequency temporal coding, whereas EAS also provides temporal coding in the low-frequency range. Patients with residual phase-locking capacities can make use of these cues.

Solid State Audio/Speech Processor Analysis.

DTIC Science & Technology

1980-03-01

techniques. The techniques were demonstrated to be worthwhile in an efficient realtime AWR system. Finally, microprocessor architectures were designed to...do not include custom chip development, detailed hardware design , construction or testing. ITTDCD is very encouraged by the results obtained in this...California, Berkley, was responsible for furnishing the simulation data of OD speech analysis techniques and for the design and development of the hardware OD

A Smartphone Application for Customized Frequency Table Selection in Cochlear Implants.

PubMed

Jethanamest, Daniel; Azadpour, Mahan; Zeman, Annette M; Sagi, Elad; Svirsky, Mario A

2017-09-01

A novel smartphone-based software application can facilitate self-selection of frequency allocation tables (FAT) in postlingually deaf cochlear implant (CI) users. CIs use FATs to represent the tonotopic organization of a normal cochlea. Current CI fitting methods typically use a standard FAT for all patients regardless of individual differences in cochlear size and electrode location. In postlingually deaf patients, different amounts of mismatch can result between the frequency-place function they experienced when they had normal hearing and the frequency-place function that results from the standard FAT. For some CI users, an alternative FAT may enhance sound quality or speech perception. Currently, no widely available tools exist to aid real-time selection of different FATs. This study aims to develop a new smartphone tool for this purpose and to evaluate speech perception and sound quality measures in a pilot study of CI subjects using this application. A smartphone application for a widely available mobile platform (iOS) was developed to serve as a preprocessor of auditory input to a clinical CI speech processor and enable interactive real-time selection of FATs. The application's output was validated by measuring electrodograms for various inputs. A pilot study was conducted in six CI subjects. Speech perception was evaluated using word recognition tests. All subjects successfully used the portable application with their clinical speech processors to experience different FATs while listening to running speech. The users were all able to select one table that they judged provided the best sound quality. All subjects chose a FAT different from the standard FAT in their everyday clinical processor. Using the smartphone application, the mean consonant-nucleus-consonant score with the default FAT selection was 28.5% (SD 16.8) and 29.5% (SD 16.4) when using a self-selected FAT. A portable smartphone application enables CI users to self-select frequency allocation tables in real time. Even though the self-selected FATs that were deemed to have better sound quality were only tested acutely (i.e., without long-term experience with them), speech perception scores were not inferior to those obtained with the clinical FATs. This software application may be a valuable tool for improving future methods of CI fitting.

Sound-direction identification, interaural time delay discrimination, and speech intelligibility advantages in noise for a bilateral cochlear implant user.

PubMed

Van Hoesel, Richard; Ramsden, Richard; Odriscoll, Martin

2002-04-01

To characterize some of the benefits available from using two cochlear implants compared with just one, sound-direction identification (ID) abilities, sensitivity to interaural time delays (ITDs) and speech intelligibility in noise were measured for a bilateral multi-channel cochlear implant user. Sound-direction ID in the horizontal plane was tested with a bilateral cochlear implant user. The subject was tested both unilaterally and bilaterally using two independent behind-the-ear ESPRIT (Cochlear Ltd.) processors, as well as bilaterally using custom research processors. Pink noise bursts were presented using an 11-loudspeaker array spanning the subject's frontal 180 degrees arc in an anechoic room. After each burst, the subject was asked to identify which loudspeaker had produced the sound. No explicit training, and no feedback were given. Presentation levels were nominally at 70 dB SPL, except for a repeat experiment using the clinical devices where the presentation levels were reduced to 60 dB SPL to avoid activation of the devices' automatic gain control (AGC) circuits. Overall presentation levels were randomly varied by +/- 3 dB. For the research processor, a "low-update-rate" and a "high-update-rate" strategy were tested. Direct measurements of ITD just noticeable differences (JNDs) were made using a 3 AFC paradigm targeting 70% correct performance on the psychometric function. Stimuli included simple, low-rate electrical pulse trains as well as high-rate pulse trains modulated at 100 Hz. Speech data comparing monaural and binaural performance in noise were also collected with both low, and high update-rate strategies on the research processors. Open-set sentences were presented from directly in front of the subject and competing multi-talker babble noise was presented from the same loudspeaker, or from a loudspeaker placed 90 degrees to the left or right of the subject. For the sound-direction ID task, monaural performance using the clinical devices showed large mean absolute errors of 81 degrees and 73 degrees, with standard deviations (averaged across all 11 loud-speakers) of 10 degrees and 17 degrees, for left and right ears, respectively. Fore bilateral device use at a presentation level of 70 dB SPL, the mean error improved to about 16 degrees with an average standard deviation of 18 degrees. When the presentation level was decreased to 60 dB SPL to avoid activation of the automatic gain control (AGC) circuits in the clinical processors, the mean response error improved further to 8 degrees with a standard deviation of 13 degrees. Further tests with the custom research processors, which had a higher stimulation rate and did not include AGCs, showed comparable response errors: around 8 or 9 degrees and a standard deviation of about 11 degrees for both update rates. The best ITD JNDs measured for this subject were between 350 to 400 microsec for simple low-rate pulse trains. Speech results showed a substantial headshadow advantage for bilateral device use when speech and noise were spatially separated, but little evidence of binaural unmasking. For spatially coincident speech and noise, listening with both ears showed similar results to listening with either side alone when loudness summation was compensated for. No significant differences were observed between binaural results for high and low update-rates in any test configuration. Only for monaural listening in one test configuration did the high rate show a small significant improvement over the low rate. Results show that even if interaural time delay cues are not well coded or perceived, bilateral implants can offer important advantages, both for speech in noise as well as for sound-direction identification.

Audiological outcomes of cochlear implantation in Waardenburg Syndrome.

PubMed

Magalhães, Ana Tereza de Matos; Samuel, Paola Angélica; Goffi-Gomez, Maria Valeria Schimdt; Tsuji, Robinson Koji; Brito, Rubens; Bento, Ricardo Ferreira

2013-07-01

 The most relevant clinical symptom in Waardenburg syndrome is profound bilateral sensorioneural hearing loss.  To characterize and describe hearing outcomes after cochlear implantation in patients with Waardenburg syndrome to improve preoperative expectations.  This was an observational and retrospective study of a series of cases. Children who were diagnosed with Waardenburg syndrome and who received a multichannel cochlear implant between March 1999 and July 2012 were included in the study. Intraoperative neural response telemetry, hearing evaluation, speech perception, and speech production data before and after surgery were assessed.  During this period, 806 patients received a cochlear implant and 10 of these (1.2%) were diagnosed with Waardenburg syndrome. Eight of the children received a Nucleus 24(®) implant and 1 child and 1 adult received a DigiSonic SP implant. The mean age at implantation was 44 months among the children. The average duration of use of a cochlear implant at the time of the study was 43 months. Intraoperative neural responses were present in all cases. Patients who could use the speech processor effectively had a pure tone average of 31 dB in free-field conditions. In addition, the MUSS and MAIS questionnaires revealed improvements in speech perception and production. Four patients did not have a good outcome, which might have been associated with ineffective use of the speech processor.  Despite the heterogeneity of the group, patients with Waardenburg syndrome who received cochlear implants were found to have hearing thresholds that allowed access to speech sounds. However, patients who received early intervention and rehabilitation showed better evolution of auditory perception.

An Intrinsically Digital Amplification Scheme for Hearing Aids

NASA Astrophysics Data System (ADS)

Blamey, Peter J.; Macfarlane, David S.; Steele, Brenton R.

2005-12-01

Results for linear and wide-dynamic range compression were compared with a new 64-channel digital amplification strategy in three separate studies. The new strategy addresses the requirements of the hearing aid user with efficient computations on an open-platform digital signal processor (DSP). The new amplification strategy is not modeled on prior analog strategies like compression and linear amplification, but uses statistical analysis of the signal to optimize the output dynamic range in each frequency band independently. Using the open-platform DSP processor also provided the opportunity for blind trial comparisons of the different processing schemes in BTE and ITE devices of a high commercial standard. The speech perception scores and questionnaire results show that it is possible to provide improved audibility for sound in many narrow frequency bands while simultaneously improving comfort, speech intelligibility in noise, and sound quality.

Development of an 8000 bps voice codec for AvSat

NASA Technical Reports Server (NTRS)

Clark, Joseph F.

1988-01-01

Air-mobile speech communication applications share robustness and noise immunity requirements with other mobile applications. The quality requirements are stringent, especially in the cockpit where air safety is involved. Based on these considerations, a decision was made to test an intermediate data rate such as 8.0 and 9.6 kb/s as proven technologies. A number of vocoders and codec technologies were investigated at rates ranging from 2.4 kb/s up to and including 9.6 kb/s. The proven vocoders operating at 2.4 and 4.8 kb/s lacked the noise immunity or the robustness to operate reliably in a cabin noise environment. One very attractive alternative approach was Spectrally Encoded Residual Excited LPC (SE-RELP) which is used in a multi-rate voice processor (MRP) developed at the Naval Research Lab (NRL). The MRP uses SE-RELP at rates of 9.6 and 16 kb/s. The 9.6 kb/s rate can be lowered to 8.0 kb/s without loss of information by modifying the frame. An 8.0 kb/s vocoder was developed using SE-RELP as a demonstrator and testbed. This demonstrator is implemented in real time using two Compaq 2 portable computers, each equipped with an ARIEL DSP016 Data Acquisition Processor.

Template Based Low Data Rate Speech Encoder

DTIC Science & Technology

1993-09-30

Nasality Distinguishes In/ from d/ 95.6 96.9 1m/ from /b/, etc. Sustention Distinguishes /f/ from /p/, $7.5 88.3 ibi from N/, Al from /0 8. etc. Sibilation...processor performs mainly Processor Workstation input/output (I/O) operations. The dynamic random access memory (DRAM) has 16 million bytes of...storage capacity. To execute the 800-b/s voice algorithm, the following amount of memory is needed: 5 MB for tables, 1.5 MB for it "program, and 30 KB for

Improving speech perception in noise for children with cochlear implants.

PubMed

Gifford, René H; Olund, Amy P; Dejong, Melissa

2011-10-01

Current cochlear implant recipients are achieving increasingly higher levels of speech recognition; however, the presence of background noise continues to significantly degrade speech understanding for even the best performers. Newer generation Nucleus cochlear implant sound processors can be programmed with SmartSound strategies that have been shown to improve speech understanding in noise for adult cochlear implant recipients. The applicability of these strategies for use in children, however, is not fully understood nor widely accepted. To assess speech perception for pediatric cochlear implant recipients in the presence of a realistic restaurant simulation generated by an eight-loudspeaker (R-SPACE™) array in order to determine whether Nucleus sound processor SmartSound strategies yield improved sentence recognition in noise for children who learn language through the implant. Single subject, repeated measures design. Twenty-two experimental subjects with cochlear implants (mean age 11.1 yr) and 25 control subjects with normal hearing (mean age 9.6 yr) participated in this prospective study. Speech reception thresholds (SRT) in semidiffuse restaurant noise originating from an eight-loudspeaker array were assessed with the experimental subjects' everyday program incorporating Adaptive Dynamic Range Optimization (ADRO) as well as with the addition of Autosensitivity control (ASC). Adaptive SRTs with the Hearing In Noise Test (HINT) sentences were obtained for all 22 experimental subjects, and performance-in percent correct-was assessed in a fixed +6 dB SNR (signal-to-noise ratio) for a six-subject subset. Statistical analysis using a repeated-measures analysis of variance (ANOVA) evaluated the effects of the SmartSound setting on the SRT in noise. The primary findings mirrored those reported previously with adult cochlear implant recipients in that the addition of ASC to ADRO significantly improved speech recognition in noise for pediatric cochlear implant recipients. The mean degree of improvement in the SRT with the addition of ASC to ADRO was 3.5 dB for a mean SRT of 10.9 dB SNR. Thus, despite the fact that these children have acquired auditory/oral speech and language through the use of their cochlear implant(s) equipped with ADRO, the addition of ASC significantly improved their ability to recognize speech in high levels of diffuse background noise. The mean SRT for the control subjects with normal hearing was 0.0 dB SNR. Given that the mean SRT for the experimental group was 10.9 dB SNR, despite the improvements in performance observed with the addition of ASC, cochlear implants still do not completely overcome the speech perception deficit encountered in noisy environments accompanying the diagnosis of severe-to-profound hearing loss. SmartSound strategies currently available in latest generation Nucleus cochlear implant sound processors are able to significantly improve speech understanding in a realistic, semidiffuse noise for pediatric cochlear implant recipients. Despite the reluctance of pediatric audiologists to utilize SmartSound settings for regular use, the results of the current study support the addition of ASC to ADRO for everyday listening environments to improve speech perception in a child's typical everyday program. American Academy of Audiology.

Noise Suppression Methods for Robust Speech Processing

DTIC Science & Technology

1981-04-01

1]. Techniques available for voice processor modification to account for noise contamination are being developed [4]. Preprocessor noise reduction...analysis window function. Principles governing discrete implementation of the transform pair are discussed, and relationships are formalized which specify

Binaural Speech Understanding With Bilateral Cochlear Implants in Reverberation.

PubMed

Kokkinakis, Kostas

2018-03-08

The purpose of this study was to investigate whether bilateral cochlear implant (CI) listeners who are fitted with clinical processors are able to benefit from binaural advantages under reverberant conditions. Another aim of this contribution was to determine whether the magnitude of each binaural advantage observed inside a highly reverberant environment differs significantly from the magnitude measured in a near-anechoic environment. Ten adults with postlingual deafness who are bilateral CI users fitted with either Nucleus 5 or Nucleus 6 clinical sound processors (Cochlear Corporation) participated in this study. Speech reception thresholds were measured in sound field and 2 different reverberation conditions (0.06 and 0.6 s) as a function of the listening condition (left, right, both) and the noise spatial location (left, front, right). The presence of the binaural effects of head-shadow, squelch, summation, and spatial release from masking in the 2 different reverberation conditions tested was determined using nonparametric statistical analysis. In the bilateral population tested, when the ambient reverberation time was equal to 0.6 s, results indicated strong positive effects of head-shadow and a weaker spatial release from masking advantage, whereas binaural squelch and summation contributed no statistically significant benefit to bilateral performance under this acoustic condition. These findings are consistent with those of previous studies, which have demonstrated that head-shadow yields the most pronounced advantage in noise. The finding that spatial release from masking produced little to almost no benefit in bilateral listeners is consistent with the hypothesis that additive reverberation degrades spatial cues and negatively affects binaural performance. The magnitude of 4 different binaural advantages was measured on the same group of bilateral CI subjects fitted with clinical processors in 2 different reverberation conditions. The results of this work demonstrate the impeding properties of reverberation on binaural speech understanding. In addition, results indicate that CI recipients who struggle in everyday listening environments are also more likely to benefit less in highly reverberant environments from their bilateral processors.

Design and Evaluation of a Personal Digital Assistant-based Research Platform for Cochlear Implants

PubMed Central

Ali, Hussnain; Lobo, Arthur P.; Loizou, Philipos C.

2014-01-01

This paper discusses the design, development, features, and clinical evaluation of a personal digital assistant (PDA)-based platform for cochlear implant research. This highly versatile and portable research platform allows researchers to design and perform complex experiments with cochlear implants manufactured by Cochlear Corporation with great ease and flexibility. The research platform includes a portable processor for implementing and evaluating novel speech processing algorithms, a stimulator unit which can be used for electrical stimulation and neurophysio-logic studies with animals, and a recording unit for collecting electroencephalogram/evoked potentials from human subjects. The design of the platform for real time and offline stimulation modes is discussed for electric-only and electric plus acoustic stimulation followed by results from an acute study with implant users for speech intelligibility in quiet and noisy conditions. The results are comparable with users’ clinical processor and very promising for undertaking chronic studies. PMID:23674422

Feasibility of through-time spiral generalized autocalibrating partial parallel acquisition for low latency accelerated real-time MRI of speech.

PubMed

Lingala, Sajan Goud; Zhu, Yinghua; Lim, Yongwan; Toutios, Asterios; Ji, Yunhua; Lo, Wei-Ching; Seiberlich, Nicole; Narayanan, Shrikanth; Nayak, Krishna S

2017-12-01

To evaluate the feasibility of through-time spiral generalized autocalibrating partial parallel acquisition (GRAPPA) for low-latency accelerated real-time MRI of speech. Through-time spiral GRAPPA (spiral GRAPPA), a fast linear reconstruction method, is applied to spiral (k-t) data acquired from an eight-channel custom upper-airway coil. Fully sampled data were retrospectively down-sampled to evaluate spiral GRAPPA at undersampling factors R = 2 to 6. Pseudo-golden-angle spiral acquisitions were used for prospective studies. Three subjects were imaged while performing a range of speech tasks that involved rapid articulator movements, including fluent speech and beat-boxing. Spiral GRAPPA was compared with view sharing, and a parallel imaging and compressed sensing (PI-CS) method. Spiral GRAPPA captured spatiotemporal dynamics of vocal tract articulators at undersampling factors ≤4. Spiral GRAPPA at 18 ms/frame and 2.4 mm 2 /pixel outperformed view sharing in depicting rapidly moving articulators. Spiral GRAPPA and PI-CS provided equivalent temporal fidelity. Reconstruction latency per frame was 14 ms for view sharing and 116 ms for spiral GRAPPA, using a single processor. Spiral GRAPPA kept up with the MRI data rate of 18ms/frame with eight processors. PI-CS required 17 minutes to reconstruct 5 seconds of dynamic data. Spiral GRAPPA enabled 4-fold accelerated real-time MRI of speech with a low reconstruction latency. This approach is applicable to wide range of speech RT-MRI experiments that benefit from real-time feedback while visualizing rapid articulator movement. Magn Reson Med 78:2275-2282, 2017. © 2017 International Society for Magnetic Resonance in Medicine. © 2017 International Society for Magnetic Resonance in Medicine.

Audiological outcomes of cochlear implantation in Waardenburg Syndrome

PubMed Central

Magalhães, Ana Tereza de Matos; Samuel, Paola Angélica; Goffi-Gomez, Maria Valeria Schimdt; Tsuji, Robinson Koji; Brito, Rubens; Bento, Ricardo Ferreira

2013-01-01

Summary Introduction: The most relevant clinical symptom in Waardenburg syndrome is profound bilateral sensorioneural hearing loss. Aim: To characterize and describe hearing outcomes after cochlear implantation in patients with Waardenburg syndrome to improve preoperative expectations. Method: This was an observational and retrospective study of a series of cases. Children who were diagnosed with Waardenburg syndrome and who received a multichannel cochlear implant between March 1999 and July 2012 were included in the study. Intraoperative neural response telemetry, hearing evaluation, speech perception, and speech production data before and after surgery were assessed. Results: During this period, 806 patients received a cochlear implant and 10 of these (1.2%) were diagnosed with Waardenburg syndrome. Eight of the children received a Nucleus 24® implant and 1 child and 1 adult received a DigiSonic SP implant. The mean age at implantation was 44 months among the children. The average duration of use of a cochlear implant at the time of the study was 43 months. Intraoperative neural responses were present in all cases. Patients who could use the speech processor effectively had a pure tone average of 31 dB in free-field conditions. In addition, the MUSS and MAIS questionnaires revealed improvements in speech perception and production. Four patients did not have a good outcome, which might have been associated with ineffective use of the speech processor. Conclusion: Despite the heterogeneity of the group, patients with Waardenburg syndrome who received cochlear implants were found to have hearing thresholds that allowed access to speech sounds. However, patients who received early intervention and rehabilitation showed better evolution of auditory perception. PMID:25992025

Speech coding at 4800 bps for mobile satellite communications

NASA Technical Reports Server (NTRS)

Gersho, Allen; Chan, Wai-Yip; Davidson, Grant; Chen, Juin-Hwey; Yong, Mei

1988-01-01

A speech compression project has recently been completed to develop a speech coding algorithm suitable for operation in a mobile satellite environment aimed at providing telephone quality natural speech at 4.8 kbps. The work has resulted in two alternative techniques which achieve reasonably good communications quality at 4.8 kbps while tolerating vehicle noise and rather severe channel impairments. The algorithms are embodied in a compact self-contained prototype consisting of two AT and T 32-bit floating-point DSP32 digital signal processors (DSP). A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. On a wirewrapped card, the prototype's circuit footprint amounts to only 200 sq cm, and consumes about 9 watts of power.

Enhancing Communication in Noisy Environments

DTIC Science & Technology

2009-10-01

derived from the ITD and ILD cues, which are binaural . ITD depends on the azimuthal position of the source. Similarly, ILD refers to the fact...4.4 dB No Perceptual Binaural Speech Enhancement [42] 4.5 dB Yes Fuzzy Cocktail Party Processor [25] 7.5 dB Yes Binaural segregation [43] 8.9 dB No...modulation. IEEE Transactions on Neural Networks. 15 (2004): 1135-50. [42] Dong R. Perceptual Binaural Speech Enhancement in Noisy Environments. M.A.Sc

Persistent Language Delay Versus Late Language Emergence in Children With Early Cochlear Implantation

PubMed Central

Nicholas, Johanna; Tobey, Emily; Davidson, Lisa

2016-01-01

Purpose The purpose of the present investigation is to differentiate children using cochlear implants (CIs) who did or did not achieve age-appropriate language scores by midelementary grades and to identify risk factors for persistent language delay following early cochlear implantation. Materials and Method Children receiving unilateral CIs at young ages (12–38 months) were tested longitudinally and classified with normal language emergence (n = 19), late language emergence (n = 22), or persistent language delay (n = 19) on the basis of their test scores at 4.5 and 10.5 years of age. Relative effects of demographic, audiological, linguistic, and academic characteristics on language emergence were determined. Results Age at CI was associated with normal language emergence but did not differentiate late emergence from persistent delay. Children with persistent delay were more likely to use left-ear implants and older speech processor technology. They experienced higher aided thresholds and lower speech perception scores. Persistent delay was foreshadowed by low morphosyntactic and phonological diversity in preschool. Logistic regression analysis predicted normal language emergence with 84% accuracy and persistent language delay with 74% accuracy. Conclusion CI characteristics had a strong effect on persistent versus resolving language delay, suggesting that right-ear (or bilateral) devices, technology upgrades, and improved audibility may positively influence long-term language outcomes. PMID:26501740

System on a chip with MPEG-4 capability

NASA Astrophysics Data System (ADS)

Yassa, Fathy; Schonfeld, Dan

2002-12-01

Current products supporting video communication applications rely on existing computer architectures. RISC processors have been used successfully in numerous applications over several decades. DSP processors have become ubiquitous in signal processing and communication applications. Real-time applications such as speech processing in cellular telephony rely extensively on the computational power of these processors. Video processors designed to implement the computationally intensive codec operations have also been used to address the high demands of video communication applications (e.g., cable set-top boxes and DVDs). This paper presents an overview of a system-on-chip (SOC) architecture used for real-time video in wireless communication applications. The SOC specifications answer to the system requirements imposed by the application environment. A CAM-based video processor is used to accelerate data intensive video compression tasks such as motion estimations and filtering. Other components are dedicated to system level data processing and audio processing. A rich set of I/Os allows the SOC to communicate with other system components such as baseband and memory subsystems.

Noise suppression methods for robust speech processing

NASA Astrophysics Data System (ADS)

Boll, S. F.; Ravindra, H.; Randall, G.; Armantrout, R.; Power, R.

1980-05-01

Robust speech processing in practical operating environments requires effective environmental and processor noise suppression. This report describes the technical findings and accomplishments during this reporting period for the research program funded to develop real time, compressed speech analysis synthesis algorithms whose performance in invariant under signal contamination. Fulfillment of this requirement is necessary to insure reliable secure compressed speech transmission within realistic military command and control environments. Overall contributions resulting from this research program include the understanding of how environmental noise degrades narrow band, coded speech, development of appropriate real time noise suppression algorithms, and development of speech parameter identification methods that consider signal contamination as a fundamental element in the estimation process. This report describes the current research and results in the areas of noise suppression using the dual input adaptive noise cancellation using the short time Fourier transform algorithms, articulation rate change techniques, and a description of an experiment which demonstrated that the spectral subtraction noise suppression algorithm can improve the intelligibility of 2400 bps, LPC 10 coded, helicopter speech by 10.6 point.

Speech Recognition of Bimodal Cochlear Implant Recipients Using a Wireless Audio Streaming Accessory for the Telephone.

PubMed

Wolfe, Jace; Morais, Mila; Schafer, Erin

2016-02-01

The goals of the present investigation were (1) to evaluate recognition of recorded speech presented over a mobile telephone for a group of adult bimodal cochlear implant users, and (2) to measure the potential benefits of wireless hearing assistance technology (HAT) for mobile telephone speech recognition using bimodal stimulation (i.e., a cochlear implant in one ear and a hearing aid on the other ear). A three-by-two-way repeated measures design was used to evaluate mobile telephone sentence-recognition performance differences obtained in quiet and in noise with and without the wireless HAT accessory coupled to the hearing aid alone, CI sound processor alone, and in the bimodal condition. Outpatient cochlear implant clinic. Sixteen bimodal users with Nucleus 24, Freedom, CI512, or CI422 cochlear implants participated in this study. Performance was measured with and without the use of a wireless HAT for the telephone used with the hearing aid alone, CI alone, and bimodal condition. CNC word recognition in quiet and in noise with and without the use of a wireless HAT telephone accessory in the hearing aid alone, CI alone, and bimodal conditions. Results suggested that the bimodal condition gave significantly better speech recognition on the mobile telephone with the wireless HAT. A wireless HAT for the mobile telephone provides bimodal users with significant improvement in word recognition in quiet and in noise over the mobile telephone.

More About Vector Adaptive/Predictive Coding Of Speech

NASA Technical Reports Server (NTRS)

Jedrey, Thomas C.; Gersho, Allen

1992-01-01

Report presents additional information about digital speech-encoding and -decoding system described in "Vector Adaptive/Predictive Encoding of Speech" (NPO-17230). Summarizes development of vector adaptive/predictive coding (VAPC) system and describes basic functions of algorithm. Describes refinements introduced enabling receiver to cope with errors. VAPC algorithm implemented in integrated-circuit coding/decoding processors (codecs). VAPC and other codecs tested under variety of operating conditions. Tests designed to reveal effects of various background quiet and noisy environments and of poor telephone equipment. VAPC found competitive with and, in some respects, superior to other 4.8-kb/s codecs and other codecs of similar complexity.

Speech recognition systems on the Cell Broadband Engine

DOE Office of Scientific and Technical Information (OSTI.GOV)

Liu, Y; Jones, H; Vaidya, S

In this paper we describe our design, implementation, and first results of a prototype connected-phoneme-based speech recognition system on the Cell Broadband Engine{trademark} (Cell/B.E.). Automatic speech recognition decodes speech samples into plain text (other representations are possible) and must process samples at real-time rates. Fortunately, the computational tasks involved in this pipeline are highly data-parallel and can receive significant hardware acceleration from vector-streaming architectures such as the Cell/B.E. Identifying and exploiting these parallelism opportunities is challenging, but also critical to improving system performance. We observed, from our initial performance timings, that a single Cell/B.E. processor can recognize speech from thousandsmore » of simultaneous voice channels in real time--a channel density that is orders-of-magnitude greater than the capacity of existing software speech recognizers based on CPUs (central processing units). This result emphasizes the potential for Cell/B.E.-based speech recognition and will likely lead to the future development of production speech systems using Cell/B.E. clusters.« less

Software reconfigurable processor technologies: the key to long-life infrastructure for future space missions

NASA Technical Reports Server (NTRS)

Srinivasan, J.; Farrington, A.; Gray, A.

2001-01-01

They present an overview of long-life reconfigurable processor technologies and of a specific architecture for implementing a software reconfigurable (software-defined) network processor for space applications.

Frequency-Limiting Effects on Speech and Environmental Sound Identification for Cochlear Implant and Normal Hearing Listeners

PubMed Central

Chang, Son-A; Won, Jong Ho; Kim, HyangHee; Oh, Seung-Ha; Tyler, Richard S.; Cho, Chang Hyun

2018-01-01

Background and Objectives It is important to understand the frequency region of cues used, and not used, by cochlear implant (CI) recipients. Speech and environmental sound recognition by individuals with CI and normal-hearing (NH) was measured. Gradients were also computed to evaluate the pattern of change in identification performance with respect to the low-pass filtering or high-pass filtering cutoff frequencies. Subjects and Methods Frequency-limiting effects were implemented in the acoustic waveforms by passing the signals through low-pass filters (LPFs) or high-pass filters (HPFs) with seven different cutoff frequencies. Identification of Korean vowels and consonants produced by a male and female speaker and environmental sounds was measured. Crossover frequencies were determined for each identification test, where the LPF and HPF conditions show the identical identification scores. Results CI and NH subjects showed changes in identification performance in a similar manner as a function of cutoff frequency for the LPF and HPF conditions, suggesting that the degraded spectral information in the acoustic signals may similarly constraint the identification performance for both subject groups. However, CI subjects were generally less efficient than NH subjects in using the limited spectral information for speech and environmental sound identification due to the inefficient coding of acoustic cues through the CI sound processors. Conclusions This finding will provide vital information in Korean for understanding how different the frequency information is in receiving speech and environmental sounds by CI processor from normal hearing. PMID:29325391

Frequency-Limiting Effects on Speech and Environmental Sound Identification for Cochlear Implant and Normal Hearing Listeners.

PubMed

Chang, Son-A; Won, Jong Ho; Kim, HyangHee; Oh, Seung-Ha; Tyler, Richard S; Cho, Chang Hyun

2017-12-01

It is important to understand the frequency region of cues used, and not used, by cochlear implant (CI) recipients. Speech and environmental sound recognition by individuals with CI and normal-hearing (NH) was measured. Gradients were also computed to evaluate the pattern of change in identification performance with respect to the low-pass filtering or high-pass filtering cutoff frequencies. Frequency-limiting effects were implemented in the acoustic waveforms by passing the signals through low-pass filters (LPFs) or high-pass filters (HPFs) with seven different cutoff frequencies. Identification of Korean vowels and consonants produced by a male and female speaker and environmental sounds was measured. Crossover frequencies were determined for each identification test, where the LPF and HPF conditions show the identical identification scores. CI and NH subjects showed changes in identification performance in a similar manner as a function of cutoff frequency for the LPF and HPF conditions, suggesting that the degraded spectral information in the acoustic signals may similarly constraint the identification performance for both subject groups. However, CI subjects were generally less efficient than NH subjects in using the limited spectral information for speech and environmental sound identification due to the inefficient coding of acoustic cues through the CI sound processors. This finding will provide vital information in Korean for understanding how different the frequency information is in receiving speech and environmental sounds by CI processor from normal hearing.

Multichannel spatial auditory display for speech communications

NASA Technical Reports Server (NTRS)

Begault, D. R.; Erbe, T.; Wenzel, E. M. (Principal Investigator)

1994-01-01

A spatial auditory display for multiple speech communications was developed at NASA/Ames Research Center. Input is spatialized by the use of simplified head-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four-letter call signs used by launch personnel at NASA against diotic speech babble. Spatial positions at 30 degrees azimuth increments were evaluated. The results from eight subjects showed a maximum intelligibility improvement of about 6-7 dB when the signal was spatialized to 60 or 90 degrees azimuth positions.

Multi-channel spatial auditory display for speech communications

NASA Astrophysics Data System (ADS)

Begault, Durand; Erbe, Tom

1993-10-01

A spatial auditory display for multiple speech communications was developed at NASA-Ames Research Center. Input is spatialized by use of simplified head-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four letter call signs used by launch personnel at NASA, against diotic speech babble. Spatial positions at 30 deg azimuth increments were evaluated. The results from eight subjects showed a maximal intelligibility improvement of about 6 to 7 dB when the signal was spatialized to 60 deg or 90 deg azimuth positions.

Multichannel spatial auditory display for speech communications.

PubMed

Begault, D R; Erbe, T

1994-10-01

A spatial auditory display for multiple speech communications was developed at NASA/Ames Research Center. Input is spatialized by the use of simplified head-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four-letter call signs used by launch personnel at NASA against diotic speech babble. Spatial positions at 30 degrees azimuth increments were evaluated. The results from eight subjects showed a maximum intelligibility improvement of about 6-7 dB when the signal was spatialized to 60 or 90 degrees azimuth positions.

Multi-channel spatial auditory display for speech communications

NASA Technical Reports Server (NTRS)

Begault, Durand; Erbe, Tom

1993-01-01

A spatial auditory display for multiple speech communications was developed at NASA-Ames Research Center. Input is spatialized by use of simplified head-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four letter call signs used by launch personnel at NASA, against diotic speech babble. Spatial positions at 30 deg azimuth increments were evaluated. The results from eight subjects showed a maximal intelligibility improvement of about 6 to 7 dB when the signal was spatialized to 60 deg or 90 deg azimuth positions.

Design of a robust baseband LPC coder for speech transmission over 9.6 kbit/s noisy channels

NASA Astrophysics Data System (ADS)

Viswanathan, V. R.; Russell, W. H.; Higgins, A. L.

1982-04-01

This paper describes the design of a baseband Linear Predictive Coder (LPC) which transmits speech over 9.6 kbit/sec synchronous channels with random bit errors of up to 1%. Presented are the results of our investigation of a number of aspects of the baseband LPC coder with the goal of maximizing the quality of the transmitted speech. Important among these aspects are: bandwidth of the baseband, coding of the baseband residual, high-frequency regeneration, and error protection of important transmission parameters. The paper discusses these and other issues, presents the results of speech-quality tests conducted during the various stages of optimization, and describes the details of the optimized speech coder. This optimized speech coding algorithm has been implemented as a real-time full-duplex system on an array processor. Informal listening tests of the real-time coder have shown that the coder produces good speech quality in the absence of channel bit errors and introduces only a slight degradation in quality for channel bit error rates of up to 1%.

DFT algorithms for bit-serial GaAs array processor architectures

NASA Technical Reports Server (NTRS)

Mcmillan, Gary B.

1988-01-01

Systems and Processes Engineering Corporation (SPEC) has developed an innovative array processor architecture for computing Fourier transforms and other commonly used signal processing algorithms. This architecture is designed to extract the highest possible array performance from state-of-the-art GaAs technology. SPEC's architectural design includes a high performance RISC processor implemented in GaAs, along with a Floating Point Coprocessor and a unique Array Communications Coprocessor, also implemented in GaAs technology. Together, these data processors represent the latest in technology, both from an architectural and implementation viewpoint. SPEC has examined numerous algorithms and parallel processing architectures to determine the optimum array processor architecture. SPEC has developed an array processor architecture with integral communications ability to provide maximum node connectivity. The Array Communications Coprocessor embeds communications operations directly in the core of the processor architecture. A Floating Point Coprocessor architecture has been defined that utilizes Bit-Serial arithmetic units, operating at very high frequency, to perform floating point operations. These Bit-Serial devices reduce the device integration level and complexity to a level compatible with state-of-the-art GaAs device technology.

Impact of a Moving Noise Masker on Speech Perception in Cochlear Implant Users

PubMed Central

Weissgerber, Tobias; Rader, Tobias; Baumann, Uwe

2015-01-01

Objectives Previous studies investigating speech perception in noise have typically been conducted with static masker positions. The aim of this study was to investigate the effect of spatial separation of source and masker (spatial release from masking, SRM) in a moving masker setup and to evaluate the impact of adaptive beamforming in comparison with fixed directional microphones in cochlear implant (CI) users. Design Speech reception thresholds (SRT) were measured in S0N0 and in a moving masker setup (S0Nmove) in 12 normal hearing participants and 14 CI users (7 subjects bilateral, 7 bimodal with a hearing aid in the contralateral ear). Speech processor settings were a moderately directional microphone, a fixed beamformer, or an adaptive beamformer. The moving noise source was generated by means of wave field synthesis and was smoothly moved in a shape of a half-circle from one ear to the contralateral ear. Noise was presented in either of two conditions: continuous or modulated. Results SRTs in the S0Nmove setup were significantly improved compared to the S0N0 setup for both the normal hearing control group and the bilateral group in continuous noise, and for the control group in modulated noise. There was no effect of subject group. A significant effect of directional sensitivity was found in the S0Nmove setup. In the bilateral group, the adaptive beamformer achieved lower SRTs than the fixed beamformer setting. Adaptive beamforming improved SRT in both CI user groups substantially by about 3 dB (bimodal group) and 8 dB (bilateral group) depending on masker type. Conclusions CI users showed SRM that was comparable to normal hearing subjects. In listening situations of everyday life with spatial separation of source and masker, directional microphones significantly improved speech perception with individual improvements of up to 15 dB SNR. Users of bilateral speech processors with both directional microphones obtained the highest benefit. PMID:25970594

Continuous Speech Recognition for Clinicians

PubMed Central

Zafar, Atif; Overhage, J. Marc; McDonald, Clement J.

1999-01-01

The current generation of continuous speech recognition systems claims to offer high accuracy (greater than 95 percent) speech recognition at natural speech rates (150 words per minute) on low-cost (under $2000) platforms. This paper presents a state-of-the-technology summary, along with insights the authors have gained through testing one such product extensively and other products superficially. The authors have identified a number of issues that are important in managing accuracy and usability. First, for efficient recognition users must start with a dictionary containing the phonetic spellings of all words they anticipate using. The authors dictated 50 discharge summaries using one inexpensive internal medicine dictionary ($30) and found that they needed to add an additional 400 terms to get recognition rates of 98 percent. However, if they used either of two more expensive and extensive commercial medical vocabularies ($349 and $695), they did not need to add terms to get a 98 percent recognition rate. Second, users must speak clearly and continuously, distinctly pronouncing all syllables. Users must also correct errors as they occur, because accuracy improves with error correction by at least 5 percent over two weeks. Users may find it difficult to train the system to recognize certain terms, regardless of the amount of training, and appropriate substitutions must be created. For example, the authors had to substitute “twice a day” for “bid” when using the less expensive dictionary, but not when using the other two dictionaries. From trials they conducted in settings ranging from an emergency room to hospital wards and clinicians' offices, they learned that ambient noise has minimal effect. Finally, they found that a minimal “usable” hardware configuration (which keeps up with dictation) comprises a 300-MHz Pentium processor with 128 MB of RAM and a “speech quality” sound card (e.g., SoundBlaster, $99). Anything less powerful will result in the system lagging behind the speaking rate. The authors obtained 97 percent accuracy with just 30 minutes of training when using the latest edition of one of the speech recognition systems supplemented by a commercial medical dictionary. This technology has advanced considerably in recent years and is now a serious contender to replace some or all of the increasingly expensive alternative methods of dictation with human transcription. PMID:10332653

Word Processing Programs and Weaker Writers/Readers: A Meta-Analysis of Research Findings

ERIC Educational Resources Information Center

Morphy, Paul; Graham, Steve

2012-01-01

Since its advent word processing has become a common writing tool, providing potential advantages over writing by hand. Word processors permit easy revision, produce legible characters quickly, and may provide additional supports (e.g., spellcheckers, speech recognition). Such advantages should remedy common difficulties among weaker…

Deep Learning-Based Noise Reduction Approach to Improve Speech Intelligibility for Cochlear Implant Recipients.

PubMed

Lai, Ying-Hui; Tsao, Yu; Lu, Xugang; Chen, Fei; Su, Yu-Ting; Chen, Kuang-Chao; Chen, Yu-Hsuan; Chen, Li-Ching; Po-Hung Li, Lieber; Lee, Chin-Hui

2018-01-20

We investigate the clinical effectiveness of a novel deep learning-based noise reduction (NR) approach under noisy conditions with challenging noise types at low signal to noise ratio (SNR) levels for Mandarin-speaking cochlear implant (CI) recipients. The deep learning-based NR approach used in this study consists of two modules: noise classifier (NC) and deep denoising autoencoder (DDAE), thus termed (NC + DDAE). In a series of comprehensive experiments, we conduct qualitative and quantitative analyses on the NC module and the overall NC + DDAE approach. Moreover, we evaluate the speech recognition performance of the NC + DDAE NR and classical single-microphone NR approaches for Mandarin-speaking CI recipients under different noisy conditions. The testing set contains Mandarin sentences corrupted by two types of maskers, two-talker babble noise, and a construction jackhammer noise, at 0 and 5 dB SNR levels. Two conventional NR techniques and the proposed deep learning-based approach are used to process the noisy utterances. We qualitatively compare the NR approaches by the amplitude envelope and spectrogram plots of the processed utterances. Quantitative objective measures include (1) normalized covariance measure to test the intelligibility of the utterances processed by each of the NR approaches; and (2) speech recognition tests conducted by nine Mandarin-speaking CI recipients. These nine CI recipients use their own clinical speech processors during testing. The experimental results of objective evaluation and listening test indicate that under challenging listening conditions, the proposed NC + DDAE NR approach yields higher intelligibility scores than the two compared classical NR techniques, under both matched and mismatched training-testing conditions. When compared to the two well-known conventional NR techniques under challenging listening condition, the proposed NC + DDAE NR approach has superior noise suppression capabilities and gives less distortion for the key speech envelope information, thus, improving speech recognition more effectively for Mandarin CI recipients. The results suggest that the proposed deep learning-based NR approach can potentially be integrated into existing CI signal processors to overcome the degradation of speech perception caused by noise.

A software tool for analyzing multichannel cochlear implant signals.

PubMed

Lai, Wai Kong; Bögli, Hans; Dillier, Norbert

2003-10-01

A useful and convenient means to analyze the radio frequency (RF) signals being sent by a speech processor to a cochlear implant would be to actually capture and display them with appropriate software. This is particularly useful for development or diagnostic purposes. sCILab (Swiss Cochlear Implant Laboratory) is such a PC-based software tool intended for the Nucleus family of Multichannel Cochlear Implants. Its graphical user interface provides a convenient and intuitive means for visualizing and analyzing the signals encoding speech information. Both numerical and graphic displays are available for detailed examination of the captured CI signals, as well as an acoustic simulation of these CI signals. sCILab has been used in the design and verification of new speech coding strategies, and has also been applied as an analytical tool in studies of how different parameter settings of existing speech coding strategies affect speech perception. As a diagnostic tool, it is also useful for troubleshooting problems with the external equipment of the cochlear implant systems.

Effect of increased IIDR in the nucleus freedom cochlear implant system.

PubMed

Holden, Laura K; Skinner, Margaret W; Fourakis, Marios S; Holden, Timothy A

2007-10-01

The objective of this study was to evaluate the effect of the increased instantaneous input dynamic range (IIDR) in the Nucleus Freedom cochlear implant (CI) system on recipients' ability to perceive soft speech and speech in noise. Ten adult Freedom CI recipients participated. Two maps differing in IIDR were placed on each subject's processor at initial activation. The IIDR was set to 30 dB for one map and 40 dB for the other. Subjects used both maps for at least one month prior to speech perception testing. Results revealed significantly higher scores for words (50 dB SPL), for sentences in background babble (65 dB SPL), and significantly lower sound field threshold levels with the 40 compared to the 30 dB IIDR map. Ceiling effects may have contributed to non-significant findings for sentences in quiet (50 dB SPL). The Freedom's increased IIDR allows better perception of soft speech and speech in noise.

VASP-4096: a very high performance programmable device for digital media processing applications

NASA Astrophysics Data System (ADS)

Krikelis, Argy

2001-03-01

Over the past few years, technology drivers for microprocessors have changed significantly. Media data delivery and processing--such as telecommunications, networking, video processing, speech recognition and 3D graphics--is increasing in importance and will soon dominate the processing cycles consumed in computer-based systems. This paper presents the architecture of the VASP-4096 processor. VASP-4096 provides high media performance with low energy consumption by integrating associative SIMD parallel processing with embedded microprocessor technology. The major innovations in the VASP-4096 is the integration of thousands of processing units in a single chip that are capable of support software programmable high-performance mathematical functions as well as abstract data processing. In addition to 4096 processing units, VASP-4096 integrates on a single chip a RISC controller that is an implementation of the SPARC architecture, 128 Kbytes of Data Memory, and I/O interfaces. The SIMD processing in VASP-4096 implements the ASProCore architecture, which is a proprietary implementation of SIMD processing, operates at 266 MHz with program instructions issued by the RISC controller. The device also integrates a 64-bit synchronous main memory interface operating at 133 MHz (double-data rate), and a 64- bit 66 MHz PCI interface. VASP-4096, compared with other processors architectures that support media processing, offers true performance scalability, support for deterministic and non-deterministic data processing on a single device, and software programmability that can be re- used in future chip generations.

Effect of Stimulation Rate on Cochlear Implant Users’ Phoneme, Word and Sentence Recognition in Quiet and in Noise

PubMed Central

Shannon, Robert V.; Cruz, Rachel J.; Galvin, John J.

2011-01-01

High stimulation rates in cochlear implants (CI) offer better temporal sampling, can induce stochastic-like firing of auditory neurons and can increase the electric dynamic range, all of which could improve CI speech performance. While commercial CI have employed increasingly high stimulation rates, no clear or consistent advantage has been shown for high rates. In this study, speech recognition was acutely measured with experimental processors in 7 CI subjects (Clarion CII users). The stimulation rate varied between (approx.) 600 and 4800 pulses per second per electrode (ppse) and the number of active electrodes varied between 4 and 16. Vowel, consonant, consonant-nucleus-consonant word and IEEE sentence recognition was acutely measured in quiet and in steady noise (+10 dB signal-to-noise ratio). Subjective quality ratings were obtained for each of the experimental processors in quiet and in noise. Except for a small difference for vowel recognition in quiet, there were no significant differences in performance among the experimental stimulation rates for any of the speech measures. There was also a small but significant increase in subjective quality rating as stimulation rates increased from 1200 to 2400 ppse in noise. Consistent with previous studies, performance significantly improved as the number of electrodes was increased from 4 to 8, but no significant difference showed between 8, 12 and 16 electrodes. Altogether, there was little-to-no advantage of high stimulation rates in quiet or in noise, at least for the present speech tests and conditions. PMID:20639631

Embedded System Implementation on FPGA System With μCLinux OS

NASA Astrophysics Data System (ADS)

Fairuz Muhd Amin, Ahmad; Aris, Ishak; Syamsul Azmir Raja Abdullah, Raja; Kalos Zakiah Sahbudin, Ratna

2011-02-01

Embedded systems are taking on more complicated tasks as the processors involved become more powerful. The embedded systems have been widely used in many areas such as in industries, automotives, medical imaging, communications, speech recognition and computer vision. The complexity requirements in hardware and software nowadays need a flexibility system for further enhancement in any design without adding new hardware. Therefore, any changes in the design system will affect the processor that need to be changed. To overcome this problem, a System On Programmable Chip (SOPC) has been designed and developed using Field Programmable Gate Array (FPGA). A softcore processor, NIOS II 32-bit RISC, which is the microprocessor core was utilized in FPGA system together with the embedded operating system(OS), μClinux. In this paper, an example of web server is explained and demonstrated

Relationships Among Peripheral and Central Electrophysiological Measures of Spatial and Spectral Selectivity and Speech Perception in Cochlear Implant Users.

PubMed

Scheperle, Rachel A; Abbas, Paul J

2015-01-01

The ability to perceive speech is related to the listener's ability to differentiate among frequencies (i.e., spectral resolution). Cochlear implant (CI) users exhibit variable speech-perception and spectral-resolution abilities, which can be attributed in part to the extent of electrode interactions at the periphery (i.e., spatial selectivity). However, electrophysiological measures of peripheral spatial selectivity have not been found to correlate with speech perception. The purpose of this study was to evaluate auditory processing at the periphery and cortex using both simple and spectrally complex stimuli to better understand the stages of neural processing underlying speech perception. The hypotheses were that (1) by more completely characterizing peripheral excitation patterns than in previous studies, significant correlations with measures of spectral selectivity and speech perception would be observed, (2) adding information about processing at a level central to the auditory nerve would account for additional variability in speech perception, and (3) responses elicited with spectrally complex stimuli would be more strongly correlated with speech perception than responses elicited with spectrally simple stimuli. Eleven adult CI users participated. Three experimental processor programs (MAPs) were created to vary the likelihood of electrode interactions within each participant. For each MAP, a subset of 7 of 22 intracochlear electrodes was activated: adjacent (MAP 1), every other (MAP 2), or every third (MAP 3). Peripheral spatial selectivity was assessed using the electrically evoked compound action potential (ECAP) to obtain channel-interaction functions for all activated electrodes (13 functions total). Central processing was assessed by eliciting the auditory change complex with both spatial (electrode pairs) and spectral (rippled noise) stimulus changes. Speech-perception measures included vowel discrimination and the Bamford-Kowal-Bench Speech-in-Noise test. Spatial and spectral selectivity and speech perception were expected to be poorest with MAP 1 (closest electrode spacing) and best with MAP 3 (widest electrode spacing). Relationships among the electrophysiological and speech-perception measures were evaluated using mixed-model and simple linear regression analyses. All electrophysiological measures were significantly correlated with each other and with speech scores for the mixed-model analysis, which takes into account multiple measures per person (i.e., experimental MAPs). The ECAP measures were the best predictor. In the simple linear regression analysis on MAP 3 data, only the cortical measures were significantly correlated with speech scores; spectral auditory change complex amplitude was the strongest predictor. The results suggest that both peripheral and central electrophysiological measures of spatial and spectral selectivity provide valuable information about speech perception. Clinically, it is often desirable to optimize performance for individual CI users. These results suggest that ECAP measures may be most useful for within-subject applications when multiple measures are performed to make decisions about processor options. They also suggest that if the goal is to compare performance across individuals based on a single measure, then processing central to the auditory nerve (specifically, cortical measures of discriminability) should be considered.

On board processor development for NASA's spaceborne imaging radar with system-on-chip technology

NASA Technical Reports Server (NTRS)

Fang, Wai-Chi

2004-01-01

This paper reports a preliminary study result of an on-board spaceborne SAR processor. It consists of a processing requirement analysis, functional specifications, and implementation with system-on-chip technology. Finally, a minimum version of this on-board processor designed for performance evaluation and for partial demonstration is illustrated.

Optimization of programming parameters in children with the advanced bionics cochlear implant.

PubMed

Baudhuin, Jacquelyn; Cadieux, Jamie; Firszt, Jill B; Reeder, Ruth M; Maxson, Jerrica L

2012-05-01

Cochlear implants provide access to soft intensity sounds and therefore improved audibility for children with severe-to-profound hearing loss. Speech processor programming parameters, such as threshold (or T-level), input dynamic range (IDR), and microphone sensitivity, contribute to the recipient's program and influence audibility. When soundfield thresholds obtained through the speech processor are elevated, programming parameters can be modified to improve soft sound detection. Adult recipients show improved detection for low-level sounds when T-levels are set at raised levels and show better speech understanding in quiet when wider IDRs are used. Little is known about the effects of parameter settings on detection and speech recognition in children using today's cochlear implant technology. The overall study aim was to assess optimal T-level, IDR, and sensitivity settings in pediatric recipients of the Advanced Bionics cochlear implant. Two experiments were conducted. Experiment 1 examined the effects of two T-level settings on soundfield thresholds and detection of the Ling 6 sounds. One program set T-levels at 10% of most comfortable levels (M-levels) and another at 10 current units (CUs) below the level judged as "soft." Experiment 2 examined the effects of IDR and sensitivity settings on speech recognition in quiet and noise. Participants were 11 children 7-17 yr of age (mean 11.3) implanted with the Advanced Bionics High Resolution 90K or CII cochlear implant system who had speech recognition scores of 20% or greater on a monosyllabic word test. Two T-level programs were compared for detection of the Ling sounds and frequency modulated (FM) tones. Differing IDR/sensitivity programs (50/0, 50/10, 70/0, 70/10) were compared using Ling and FM tone detection thresholds, CNC (consonant-vowel nucleus-consonant) words at 50 dB SPL, and Hearing in Noise Test for Children (HINT-C) sentences at 65 dB SPL in the presence of four-talker babble (+8 signal-to-noise ratio). Outcomes were analyzed using a paired t-test and a mixed-model repeated measures analysis of variance (ANOVA). T-levels set 10 CUs below "soft" resulted in significantly lower detection thresholds for all six Ling sounds and FM tones at 250, 1000, 3000, 4000, and 6000 Hz. When comparing programs differing by IDR and sensitivity, a 50 dB IDR with a 0 sensitivity setting showed significantly poorer thresholds for low frequency FM tones and voiced Ling sounds. Analysis of group mean scores for CNC words in quiet or HINT-C sentences in noise indicated no significant differences across IDR/sensitivity settings. Individual data, however, showed significant differences between IDR/sensitivity programs in noise; the optimal program differed across participants. In pediatric recipients of the Advanced Bionics cochlear implant device, manually setting T-levels with ascending loudness judgments should be considered when possible or when low-level sounds are inaudible. Study findings confirm the need to determine program settings on an individual basis as well as the importance of speech recognition verification measures in both quiet and noise. Clinical guidelines are suggested for selection of programming parameters in both young and older children. American Academy of Audiology.

Baseband processor development for the Advanced Communications Satellite Program

NASA Technical Reports Server (NTRS)

Moat, D.; Sabourin, D.; Stilwell, J.; Mccallister, R.; Borota, M.

1982-01-01

An onboard-baseband-processor concept for a satellite-switched time-division-multiple-access (SS-TDMA) communication system was developed for NASA Lewis Research Center. The baseband processor routes and controls traffic on an individual message basis while providing significant advantages in improved link margins and system flexibility. Key technology developments required to prove the flight readiness of the baseband-processor design are being verified in a baseband-processor proof-of-concept model. These technology developments include serial MSK modems, Clos-type baseband routing switch, a single-chip CMOS maximum-likelihood convolutional decoder, and custom LSL implementation of high-speed, low-power ECL building blocks.

Compressed Speech Technology: Implications for Learning and Instruction.

ERIC Educational Resources Information Center

Sullivan, LeRoy L.

This paper first traces the historical development of speech compression technology, which has made it possible to alter the spoken rate of a pre-recorded message without excessive distortion. Terms used to describe techniques employed as the technology evolved are discussed, including rapid speech, rate altered speech, cut-and-spliced speech, and…

Relationships Among Peripheral and Central Electrophysiological Measures of Spatial and Spectral Selectivity and Speech Perception in Cochlear Implant Users

PubMed Central

Scheperle, Rachel A.; Abbas, Paul J.

2014-01-01

Objectives The ability to perceive speech is related to the listener’s ability to differentiate among frequencies (i.e., spectral resolution). Cochlear implant (CI) users exhibit variable speech-perception and spectral-resolution abilities, which can be attributed in part to the extent of electrode interactions at the periphery (i.e., spatial selectivity). However, electrophysiological measures of peripheral spatial selectivity have not been found to correlate with speech perception. The purpose of this study was to evaluate auditory processing at the periphery and cortex using both simple and spectrally complex stimuli to better understand the stages of neural processing underlying speech perception. The hypotheses were that (1) by more completely characterizing peripheral excitation patterns than in previous studies, significant correlations with measures of spectral selectivity and speech perception would be observed, (2) adding information about processing at a level central to the auditory nerve would account for additional variability in speech perception, and (3) responses elicited with spectrally complex stimuli would be more strongly correlated with speech perception than responses elicited with spectrally simple stimuli. Design Eleven adult CI users participated. Three experimental processor programs (MAPs) were created to vary the likelihood of electrode interactions within each participant. For each MAP, a subset of 7 of 22 intracochlear electrodes was activated: adjacent (MAP 1), every-other (MAP 2), or every third (MAP 3). Peripheral spatial selectivity was assessed using the electrically evoked compound action potential (ECAP) to obtain channel-interaction functions for all activated electrodes (13 functions total). Central processing was assessed by eliciting the auditory change complex (ACC) with both spatial (electrode pairs) and spectral (rippled noise) stimulus changes. Speech-perception measures included vowel-discrimination and the Bamford-Kowal-Bench Sentence-in-Noise (BKB-SIN) test. Spatial and spectral selectivity and speech perception were expected to be poorest with MAP 1 (closest electrode spacing) and best with MAP 3 (widest electrode spacing). Relationships among the electrophysiological and speech-perception measures were evaluated using mixed-model and simple linear regression analyses. Results All electrophysiological measures were significantly correlated with each other and with speech perception for the mixed-model analysis, which takes into account multiple measures per person (i.e. experimental MAPs). The ECAP measures were the best predictor of speech perception. In the simple linear regression analysis on MAP 3 data, only the cortical measures were significantly correlated with speech; spectral ACC amplitude was the strongest predictor. Conclusions The results suggest that both peripheral and central electrophysiological measures of spatial and spectral selectivity provide valuable information about speech perception. Clinically, it is often desirable to optimize performance for individual CI users. These results suggest that ECAP measures may be the most useful for within-subject applications, when multiple measures are performed to make decisions about processor options. They also suggest that if the goal is to compare performance across individuals based on single measure, then processing central to the auditory nerve (specifically, cortical measures of discriminability) should be considered. PMID:25658746

Understanding Language: An Information-Processing Analysis of Speech Perception, Reading, and Psycholinguistics.

ERIC Educational Resources Information Center

Massaro, Dominic W., Ed.

In an information-processing approach to language processing, language processing is viewed as a sequence of psychological stages that occur between the initial presentation of the language stimulus and the meaning in the mind of the language processor. This book defines each of the processes and structures involved, explains how each of them…

Application of Advanced Multi-Core Processor Technologies to Oceanographic Research

DTIC Science & Technology

2013-09-30

STM32 NXP LPC series No Proprietary Microchip PIC32/DSPIC No > 500 mW; < 5 W ARM Cortex TI OMAP TI Sitara Broadcom BCM2835 Varies FPGA...1 DISTRIBUTION STATEMENT A. Approved for public release; distribution is unlimited. Application of Advanced Multi-Core Processor Technologies...state-of-the-art information processing architectures. OBJECTIVES Next-generation processor architectures (multi-core, multi-threaded) hold the

Talking Wheelchair

NASA Technical Reports Server (NTRS)

1981-01-01

Communication is made possible for disabled individuals by means of an electronic system, developed at Stanford University's School of Medicine, which produces highly intelligible synthesized speech. Familiarly known as the "talking wheelchair" and formally as the Versatile Portable Speech Prosthesis (VPSP). Wheelchair mounted system consists of a word processor, a video screen, a voice synthesizer and a computer program which instructs the synthesizer how to produce intelligible sounds in response to user commands. Computer's memory contains 925 words plus a number of common phrases and questions. Memory can also store several thousand other words of the user's choice. Message units are selected by operating a simple switch, joystick or keyboard. Completed message appears on the video screen, then user activates speech synthesizer, which generates a voice with a somewhat mechanical tone. With the keyboard, an experienced user can construct messages as rapidly as 30 words per minute.

Onboard processor technology review

NASA Technical Reports Server (NTRS)

Benz, Harry F.

1990-01-01

The general need and requirements for the onboard embedded processors necessary to control and manipulate data in spacecraft systems are discussed. The current known requirements are reviewed from a user perspective, based on current practices in the spacecraft development process. The current capabilities of available processor technologies are then discussed, and these are projected to the generation of spacecraft computers currently under identified, funded development. An appraisal is provided for the current national developmental effort.

Use of Computer Speech Technologies To Enhance Learning.

ERIC Educational Resources Information Center

Ferrell, Joe

1999-01-01

Discusses the design of an innovative learning system that uses new technologies for the man-machine interface, incorporating a combination of Automatic Speech Recognition (ASR) and Text To Speech (TTS) synthesis. Highlights include using speech technologies to mimic the attributes of the ideal tutor and design features. (AEF)

Speech systems research at Texas Instruments

NASA Technical Reports Server (NTRS)

Doddington, George R.

1977-01-01

An assessment of automatic speech processing technology is presented. Fundamental problems in the development and the deployment of automatic speech processing systems are defined and a technology forecast for speech systems is presented.

JPRS Report, Science & Technology, Europe.

DTIC Science & Technology

1991-04-30

processor in collaboration with Intel . The processor , christened Touchstone, will be used as the core of a parallel computer with 2,000 processors . One of...ELECTRONIQUE HEBDO in French 24 Jan 91 pp 14-15 [Article by Claire Remy: "Everything Set for Neural Signal Processors " first paragraph is ELECTRONIQUE...paving the way for neural signal processors in so doing. The principal advantage of this specific circuit over a neuromimetic software program is

Sperry Univac speech communications technology

NASA Technical Reports Server (NTRS)

Medress, Mark F.

1977-01-01

Technology and systems for effective verbal communication with computers were developed. A continuous speech recognition system for verbal input, a word spotting system to locate key words in conversational speech, prosodic tools to aid speech analysis, and a prerecorded voice response system for speech output are described.

Advanced Avionics and Processor Systems for a Flexible Space Exploration Architecture

NASA Technical Reports Server (NTRS)

Keys, Andrew S.; Adams, James H.; Smith, Leigh M.; Johnson, Michael A.; Cressler, John D.

2010-01-01

The Advanced Avionics and Processor Systems (AAPS) project, formerly known as the Radiation Hardened Electronics for Space Environments (RHESE) project, endeavors to develop advanced avionic and processor technologies anticipated to be used by NASA s currently evolving space exploration architectures. The AAPS project is a part of the Exploration Technology Development Program, which funds an entire suite of technologies that are aimed at enabling NASA s ability to explore beyond low earth orbit. NASA s Marshall Space Flight Center (MSFC) manages the AAPS project. AAPS uses a broad-scoped approach to developing avionic and processor systems. Investment areas include advanced electronic designs and technologies capable of providing environmental hardness, reconfigurable computing techniques, software tools for radiation effects assessment, and radiation environment modeling tools. Near-term emphasis within the multiple AAPS tasks focuses on developing prototype components using semiconductor processes and materials (such as Silicon-Germanium (SiGe)) to enhance a device s tolerance to radiation events and low temperature environments. As the SiGe technology will culminate in a delivered prototype this fiscal year, the project emphasis shifts its focus to developing low-power, high efficiency total processor hardening techniques. In addition to processor development, the project endeavors to demonstrate techniques applicable to reconfigurable computing and partially reconfigurable Field Programmable Gate Arrays (FPGAs). This capability enables avionic architectures the ability to develop FPGA-based, radiation tolerant processor boards that can serve in multiple physical locations throughout the spacecraft and perform multiple functions during the course of the mission. The individual tasks that comprise AAPS are diverse, yet united in the common endeavor to develop electronics capable of operating within the harsh environment of space. Specifically, the AAPS tasks for the Federal fiscal year of 2010 are: Silicon-Germanium (SiGe) Integrated Electronics for Extreme Environments, Modeling of Radiation Effects on Electronics, Radiation Hardened High Performance Processors (HPP), and and Reconfigurable Computing.

Effects of hearing aid settings for electric-acoustic stimulation.

PubMed

Dillon, Margaret T; Buss, Emily; Pillsbury, Harold C; Adunka, Oliver F; Buchman, Craig A; Adunka, Marcia C

2014-02-01

Cochlear implant (CI) recipients with postoperative hearing preservation may utilize an ipsilateral bimodal listening condition known as electric-acoustic stimulation (EAS). Studies on EAS have reported significant improvements in speech perception abilities over CI-alone listening conditions. Adjustments to the hearing aid (HA) settings to match prescription targets routinely used in the programming of conventional amplification may provide additional gains in speech perception abilities. Investigate the difference in users' speech perception scores when listening with the recommended HA settings for EAS patients versus HA settings adjusted to match National Acoustic Laboratories' nonlinear fitting procedure version 1 (NAL-NL1) targets. Prospective analysis of the influence of HA settings. Nine EAS recipients with greater than 12 mo of listening experience with the DUET speech processor. Subjects were tested in the EAS listening condition with two different HA setting configurations. Speech perception materials included consonant-nucleus-consonant (CNC) words in quiet, AzBio sentences in 10-talker speech babble at a signal-to-noise ratio (SNR) of +10, and the Bamford-Kowal-Bench sentences in noise (BKB-SIN) test. The speech perception performance on each test measure was compared between the two HA configurations. Subjects experienced a significant improvement in speech perception abilities with the HA settings adjusted to match NAL-NL1 targets over the recommended HA settings. EAS subjects have been shown to experience improvements in speech perception abilities when listening to ipsilateral combined stimulation. This population's abilities may be underestimated with current HA settings. Tailoring the HA output to the patient's individual hearing loss offers improved outcomes on speech perception measures. American Academy of Audiology.

Embedded Data Processor and Portable Computer Technology testbeds

NASA Technical Reports Server (NTRS)

Alena, Richard; Liu, Yuan-Kwei; Goforth, Andre; Fernquist, Alan R.

1993-01-01

Attention is given to current activities in the Embedded Data Processor and Portable Computer Technology testbed configurations that are part of the Advanced Data Systems Architectures Testbed at the Information Sciences Division at NASA Ames Research Center. The Embedded Data Processor Testbed evaluates advanced microprocessors for potential use in mission and payload applications within the Space Station Freedom Program. The Portable Computer Technology (PCT) Testbed integrates and demonstrates advanced portable computing devices and data system architectures. The PCT Testbed uses both commercial and custom-developed devices to demonstrate the feasibility of functional expansion and networking for portable computers in flight missions.

Advanced Avionics and Processor Systems for Space and Lunar Exploration

NASA Technical Reports Server (NTRS)

Keys, Andrew S.; Adams, James H.; Ray, Robert E.; Johnson, Michael A.; Cressler, John D.

2009-01-01

NASA's newly named Advanced Avionics and Processor Systems (AAPS) project, formerly known as the Radiation Hardened Electronics for Space Environments (RHESE) project, endeavors to mature and develop the avionic and processor technologies required to fulfill NASA's goals for future space and lunar exploration. Over the past year, multiple advancements have been made within each of the individual AAPS technology development tasks that will facilitate the success of the Constellation program elements. This paper provides a brief review of the project's recent technology advancements, discusses their application to Constellation projects, and addresses the project's plans for the coming year.

Application of advanced speech technology in manned penetration bombers

NASA Astrophysics Data System (ADS)

North, R.; Lea, W.

1982-03-01

This report documents research on the potential use of speech technology in a manned penetration bomber aircraft (B-52/G and H). The objectives of the project were to analyze the pilot/copilot crewstation tasks over a three-hour-and forty-minute mission and determine the tasks that would benefit the most from conversion to speech recognition/generation, determine the technological feasibility of each of the identified tasks, and prioritize these tasks based on these criteria. Secondary objectives of the program were to enunciate research strategies in the application of speech technologies in airborne environments, and develop guidelines for briefing user commands on the potential of using speech technologies in the cockpit. The results of this study indicated that for the B-52 crewmember, speech recognition would be most beneficial for retrieving chart and procedural data that is contained in the flight manuals. Technological feasibility of these tasks indicated that the checklist and procedural retrieval tasks would be highly feasible for a speech recognition system.

Simulating Synchronous Processors

DTIC Science & Technology

1988-06-01

34f Fvtvru m LABORATORY FOR INMASSACHUSETTSFCOMPUTER SCIENCE TECHNOLOGY MIT/LCS/TM-359 SIMULATING SYNCHRONOUS PROCESSORS Jennifer Lundelius Welch...PROJECT TASK WORK UNIT Arlington, VA 22217 ELEMENT NO. NO. NO ACCESSION NO. 11. TITLE Include Security Classification) Simulating Synchronous Processors...necessary and identify by block number) In this paper we show how a distributed system with synchronous processors and asynchro- nous message delays can

Technology Developments in Radiation-Hardened Electronics for Space Environments

NASA Technical Reports Server (NTRS)

Keys, Andrew S.; Howell, Joe T.

2008-01-01

The Radiation Hardened Electronics for Space Environments (RHESE) project consists of a series of tasks designed to develop and mature a broad spectrum of radiation hardened and low temperature electronics technologies. Three approaches are being taken to address radiation hardening: improved material hardness, design techniques to improve radiation tolerance, and software methods to improve radiation tolerance. Within these approaches various technology products are being addressed including Field Programmable Gate Arrays (FPGA), Field Programmable Analog Arrays (FPAA), MEMS, Serial Processors, Reconfigurable Processors, and Parallel Processors. In addition to radiation hardening, low temperature extremes are addressed with a focus on material and design approaches. System level applications for the RHESE technology products are discussed.

Voice synthesis application

NASA Astrophysics Data System (ADS)

Lightstone, P. C.; Davidson, W. M.

1982-04-01

The military detection assessment laboratory houses an experimental field system which assesses different alarm indicators such as fence disturbance sensors, MILES cables, and microwave Racons. A speech synthesis board which could be interfaced, by means of a computer, to an alarm logger making verbal acknowledgement of alarms possible was purchased. Different products and different types of voice synthesis were analyzed before a linear predictive code device produced by Telesensory Speech Systems of Palo Alto, California was chosen. This device is called the Speech 1000 Board and has a dedicated 8085 processor. A multiplexer card was designed and the Sp 1000 interfaced through the card into a TMS 990/100M Texas Instrument microcomputer. It was also necessary to design the software with the capability of recognizing and flagging an alarm on any 1 of 32 possible lines. The experimental field system was then packaged with a dc power supply, LED indicators, speakers, and switches, and deployed in the field performing reliably.

Fast particles identification in programmable form at level-0 trigger by means of the 3D-Flow system

DOE Office of Scientific and Technical Information (OSTI.GOV)

Crosetto, Dario B.

1998-10-30

The 3D-Flow Processor system is a new, technology-independent concept in very fast, real-time system architectures. Based on either an FPGA or an ASIC implementation, it can address, in a fully programmable manner, applications where commercially available processors would fail because of throughput requirements. Possible applications include filtering-algorithms (pattern recognition) from the input of multiple sensors, as well as moving any input validated by these filtering-algorithms to a single output channel. Both operations can easily be implemented on a 3D-Flow system to achieve a real-time processing system with a very short lag time. This system can be built either with off-the-shelfmore » FPGAs or, for higher data rates, with CMOS chips containing 4 to 16 processors each. The basic building block of the system, a 3D-Flow processor, has been successfully designed in VHDL code written in ''Generic HDL'' (mostly made of reusable blocks that are synthesizable in different technologies, or FPGAs), to produce a netlist for a four-processor ASIC featuring 0.35 micron CBA (Ceil Base Array) technology at 3.3 Volts, 884 mW power dissipation at 60 MHz and 63.75 mm sq. die size. The same VHDL code has been targeted to three FPGA manufacturers (Altera EPF10K250A, ORCA-Lucent Technologies 0R3T165 and Xilinx XCV1000). A complete set of software tools, the 3D-Flow System Manager, equally applicable to ASIC or FPGA implementations, has been produced to provide full system simulation, application development, real-time monitoring, and run-time fault recovery. Today's technology can accommodate 16 processors per chip in a medium size die, at a cost per processor of less than $5 based on the current silicon die/size technology cost.« less

Miniature Fuel Processors for Portable Fuel Cell Power Supplies

DOE Office of Scientific and Technical Information (OSTI.GOV)

Holladay, Jamie D.; Jones, Evan O.; Palo, Daniel R.

2003-06-02

Miniature and micro-scale fuel processors are discussed. The enabling technologies for these devices are the novel catalysts and the micro-technology-based designs. The novel catalyst allows for methanol reforming at high gas hourly space velocities of 50,000 hr-1 or higher, while maintaining a carbon monoxide levels at 1% or less. The micro-technology-based designs enable the devices to be extremely compact and lightweight. The miniature fuel processors can nominally provide between 25-50 watts equivalent of hydrogen which is ample for soldier or personal portable power supplies. The integrated processors have a volume less than 50 cm3, a mass less than 150 grams,more » and thermal efficiencies of up to 83%. With reasonable assumptions on fuel cell efficiencies, anode gas and water management, parasitic power loss, etc., the energy density was estimated at 1700 Whr/kg. The miniature processors have been demonstrated with a carbon monoxide clean-up method and a fuel cell stack. The micro-scale fuel processors have been designed to provide up to 0.3 watt equivalent of power with efficiencies over 20%. They have a volume of less than 0.25 cm3 and a mass of less than 1 gram.« less

[The Russian-language version of the matrix test (RUMatrix) in free field in patients after cochlear implantation in the long term].

PubMed

Goykhburg, M V; Bakhshinyan, V V; Petrova, I P; Wazybok, A; Kollmeier, B; Tavartkiladze, G A

The deterioration of speech intelligibility in the patients using cochlear implantation (CI) systems is especially well apparent in the noisy environment. It explains why phrasal speech tests, such as a Matrix sentence test, have become increasingly more popular in the speech audiometry during rehabilitation after CI. The Matrix test allows to estimate speech perception by the patients in a real life situation. The objective of this study was to assess the effectiveness of audiological rehabilitation of CI patients using the Russian-language version of the matrix test (RUMatrix) in free field in the noisy environment. 33 patients aged from 5 to 40 years with a more than 3 year experience of using cochlear implants inserted at the National Research Center for Audiology and Hearing Rehabilitation were included in our study. Five of these patients were implanted bilaterally. The results of our study showed a statistically significant improvement of speech intelligibility in the noisy environment after the speech processor adjustment; dynamics of the signal-to-noise ratio changes was -1.7 dB (p<0.001). The RUMatrix test is a highly efficient method for the estimation of speech intelligibility in the patients undergoing clinical investigations in the noisy environment. The high degree of comparability of the RUMatrix test with the Matrix tests in other languages makes possible its application in international multicenter studies.

Auditory cortical activation and plasticity after cochlear implantation measured by PET using fluorodeoxyglucose.

PubMed

Łukaszewicz-Moszyńska, Zuzanna; Lachowska, Magdalena; Niemczyk, Kazimierz

2014-01-01

The purpose of this study was to evaluate possible relationships between duration of cochlear implant use and results of positron emission tomography (PET) measurements in the temporal lobes performed while subjects listened to speech stimuli. Other aspects investigated were whether implantation side impacts significantly on cortical representations of functions related to understanding speech (ipsi- or contralateral to the implanted side) and whether any correlation exists between cortical activation and speech therapy results. Objective cortical responses to acoustic stimulation were measured, using PET, in nine cochlear implant patients (age range: 15 to 50 years). All the patients suffered from bilateral deafness, were right-handed, and had no additional neurological deficits. They underwent PET imaging three times: immediately after the first fitting of the speech processor (activation of the cochlear implant), and one and two years later. A tendency towards increasing levels of activation in areas of the primary and secondary auditory cortex on the left side of the brain was observed. There was no clear effect of the side of implantation (left or right) on the degree of cortical activation in the temporal lobe. However, the PET results showed a correlation between degree of cortical activation and speech therapy results.

Auditory cortical activation and plasticity after cochlear implantation measured by PET using fluorodeoxyglucose

PubMed Central

Łukaszewicz-Moszyńska, Zuzanna; Lachowska, Magdalena; Niemczyk, Kazimierz

2014-01-01

Summary The purpose of this study was to evaluate possible relationships between duration of cochlear implant use and results of positron emission tomography (PET) measurements in the temporal lobes performed while subjects listened to speech stimuli. Other aspects investigated were whether implantation side impacts significantly on cortical representations of functions related to understanding speech (ipsi- or contralateral to the implanted side) and whether any correlation exists between cortical activation and speech therapy results. Objective cortical responses to acoustic stimulation were measured, using PET, in nine cochlear implant patients (age range: 15 to 50 years). All the patients suffered from bilateral deafness, were right-handed, and had no additional neurological deficits. They underwent PET imaging three times: immediately after the first fitting of the speech processor (activation of the cochlear implant), and one and two years later. A tendency towards increasing levels of activation in areas of the primary and secondary auditory cortex on the left side of the brain was observed. There was no clear effect of the side of implantation (left or right) on the degree of cortical activation in the temporal lobe. However, the PET results showed a correlation between degree of cortical activation and speech therapy results. PMID:25306122

Perceptual Learning and Auditory Training in Cochlear Implant Recipients

PubMed Central

Fu, Qian-Jie; Galvin, John J.

2007-01-01

Learning electrically stimulated speech patterns can be a new and difficult experience for cochlear implant (CI) recipients. Recent studies have shown that most implant recipients at least partially adapt to these new patterns via passive, daily-listening experiences. Gradually introducing a speech processor parameter (eg, the degree of spectral mismatch) may provide for more complete and less stressful adaptation. Although the implant device restores hearing sensation and the continued use of the implant provides some degree of adaptation, active auditory rehabilitation may be necessary to maximize the benefit of implantation for CI recipients. Currently, there are scant resources for auditory rehabilitation for adult, postlingually deafened CI recipients. We recently developed a computer-assisted speech-training program to provide the means to conduct auditory rehabilitation at home. The training software targets important acoustic contrasts among speech stimuli, provides auditory and visual feedback, and incorporates progressive training techniques, thereby maintaining recipients’ interest during the auditory training exercises. Our recent studies demonstrate the effectiveness of targeted auditory training in improving CI recipients’ speech and music perception. Provided with an inexpensive and effective auditory training program, CI recipients may find the motivation and momentum to get the most from the implant device. PMID:17709574

Effect of technological advances on cochlear implant performance in adults.

PubMed

Lenarz, Minoo; Joseph, Gert; Sönmez, Hasibe; Büchner, Andreas; Lenarz, Thomas

2011-12-01

To evaluate the effect of technological advances in the past 20 years on the hearing performance of a large cohort of adult cochlear implant (CI) patients. Individual, retrospective, cohort study. According to technological developments in electrode design and speech-processing strategies, we defined five virtual intervals on the time scale between 1984 and 2008. A cohort of 1,005 postlingually deafened adults was selected for this study, and their hearing performance with a CI was evaluated retrospectively according to these five technological intervals. The test battery was composed of four standard German speech tests: Freiburger monosyllabic test, speech tracking test, Hochmair-Schulz-Moser (HSM) sentence test in quiet, and HSM sentence test in 10 dB noise. The direct comparison of the speech perception in postlingually deafened adults, who were implanted during different technological periods, reveals an obvious improvement in the speech perception in patients who benefited from the recent electrode designs and speech-processing strategies. The major influence of technological advances on CI performance seems to be on speech perception in noise. Better speech perception in noisy surroundings is strong proof for demonstrating the success rate of new electrode designs and speech-processing strategies. Standard (internationally comparable) speech tests in noise should become an obligatory part of the postoperative test battery for adult CI patients. Copyright © 2011 The American Laryngological, Rhinological, and Otological Society, Inc.

Multichannel Spatial Auditory Display for Speed Communications

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Erbe, Tom

1994-01-01

A spatial auditory display for multiple speech communications was developed at NASA/Ames Research Center. Input is spatialized by the use of simplifiedhead-related transfer functions, adapted for FIR filtering on Motorola 56001 digital signal processors. Hardware and firmware design implementations are overviewed for the initial prototype developed for NASA-Kennedy Space Center. An adaptive staircase method was used to determine intelligibility levels of four-letter call signs used by launch personnel at NASA against diotic speech babble. Spatial positions at 30 degree azimuth increments were evaluated. The results from eight subjects showed a maximum intelligibility improvement of about 6-7 dB when the signal was spatialized to 60 or 90 degree azimuth positions.

Processor architecture for airborne SAR systems

NASA Technical Reports Server (NTRS)

Glass, C. M.

1983-01-01

Digital processors for spaceborne imaging radars and application of the technology developed for airborne SAR systems are considered. Transferring algorithms and implementation techniques from airborne to spaceborne SAR processors offers obvious advantages. The following topics are discussed: (1) a quantification of the differences in processing algorithms for airborne and spaceborne SARs; and (2) an overview of three processors for airborne SAR systems.

Optoelectronic Technology Consortium: Precompetitive Consortium for Optoelectronic Interconnect Technology

DTIC Science & Technology

1992-09-01

demonstrating the producibility of optoelectronic components for high-density/high-data-rate processors and accelerating the insertion of this technology...technology development stage, OETC will advance the development of optical components, produce links for a multiboard processor testbed demonstration, and...components that are affordable, initially at <$100 per line, and reliable, with a li~e BER᝺-15 and MTTF >10 6 hours. Under the OETC program, Honeywell will

Ultra-Reliable Digital Avionics (URDA) processor

NASA Astrophysics Data System (ADS)

Branstetter, Reagan; Ruszczyk, William; Miville, Frank

1994-10-01

Texas Instruments Incorporated (TI) developed the URDA processor design under contract with the U.S. Air Force Wright Laboratory and the U.S. Army Night Vision and Electro-Sensors Directorate. TI's approach couples advanced packaging solutions with advanced integrated circuit (IC) technology to provide a high-performance (200 MIPS/800 MFLOPS) modular avionics processor module for a wide range of avionics applications. TI's processor design integrates two Ada-programmable, URDA basic processor modules (BPM's) with a JIAWG-compatible PiBus and TMBus on a single F-22 common integrated processor-compatible form-factor SEM-E avionics card. A separate, high-speed (25-MWord/second 32-bit word) input/output bus is provided for sensor data. Each BPM provides a peak throughput of 100 MIPS scalar concurrent with 400-MFLOPS vector processing in a removable multichip module (MCM) mounted to a liquid-flowthrough (LFT) core and interfacing to a processor interface module printed wiring board (PWB). Commercial RISC technology coupled with TI's advanced bipolar complementary metal oxide semiconductor (BiCMOS) application specific integrated circuit (ASIC) and silicon-on-silicon packaging technologies are used to achieve the high performance in a miniaturized package. A Mips R4000-family reduced instruction set computer (RISC) processor and a TI 100-MHz BiCMOS vector coprocessor (VCP) ASIC provide, respectively, the 100 MIPS of a scalar processor throughput and 400 MFLOPS of vector processing throughput for each BPM. The TI Aladdim ASIC chipset was developed on the TI Aladdin Program under contract with the U.S. Army Communications and Electronics Command and was sponsored by the Advanced Research Projects Agency with technical direction from the U.S. Army Night Vision and Electro-Sensors Directorate.

Selection of a Brine Processor Technology for NASA Manned Missions

NASA Technical Reports Server (NTRS)

Carter, Donald L.; Gleich, Andrew F.

2016-01-01

The current ISS Water Recovery System (WRS) reclaims water from crew urine, humidity condensate, and Sabatier product water. Urine is initially processed by the Urine Processor Assembly (UPA) which recovers 75% of the urine as distillate. The remainder of the water is present in the waste brine which is currently disposed of as trash on ISS. For future missions this additional water must be reclaimed due to the significant resupply penalty for missions beyond Low Earth Orbit (LEO). NASA has pursued various technology development programs for a brine processor in the past several years. This effort has culminated in a technology down-select to identify the optimum technology for future manned missions. The technology selection is based on various criteria, including mass, power, reliability, maintainability, and safety. Beginning in 2016 the selected technology will be transitioned to a flight hardware program for demonstration on ISS. This paper summarizes the technology selection process, the competing technologies, and the rationale for the technology selected for future manned missions.

Technologies for the Study of Speech: Review and an Application

ERIC Educational Resources Information Center

Babatsouli, Elena

2015-01-01

Technologies used for the study of speech are classified here into non-intrusive and intrusive. The paper informs on current non-intrusive technologies that are used for linguistic investigations of the speech signal, both phonological and phonetic. Providing a point of reference, the review covers existing technological advances in language…

DEMONSTRATION BULLETIN: AOSTRA-SOILTECH ANAEROBIC THERMAL PROCESSOR: WIDE BEACH DEVELOPMENT SITE

EPA Science Inventory

The anaerobic thermal processor (ATP) was developed by UMATAC Industrial Processes under the sponsorship of the Alberta Oil Sands Technology and Research Authority (AOSTRA) and is licensed by SoilTech ATP Systems, Inc., a U.S. corporation. The ATP technology involves a physi...

Potential Benefits of an Integrated Electric-Acoustic Sound Processor with Children: A Preliminary Report.

PubMed

Wolfe, Jace; Neumann, Sara; Schafer, Erin; Marsh, Megan; Wood, Mark; Baker, R Stanley

2017-02-01

A number of published studies have demonstrated the benefits of electric-acoustic stimulation (EAS) over conventional electric stimulation for adults with functional low-frequency acoustic hearing and severe-to-profound high-frequency hearing loss. These benefits potentially include better speech recognition in quiet and in noise, better localization, improvements in sound quality, better music appreciation and aptitude, and better pitch recognition. There is, however, a paucity of published reports describing the potential benefits and limitations of EAS for children with functional low-frequency acoustic hearing and severe-to-profound high-frequency hearing loss. The objective of this study was to explore the potential benefits of EAS for children. A repeated measures design was used to evaluate performance differences obtained with EAS stimulation versus acoustic- and electric-only stimulation. Seven users of Cochlear Nucleus Hybrid, Nucleus 24 Freedom, CI512, and CI422 implants were included in the study. Sentence recognition (assayed using the pediatric version of the AzBio sentence recognition test) was evaluated in quiet and at three fixed signal-to-noise ratios (SNR) (0, +5, and +10 dB). Functional hearing performance was also evaluated with the use of questionnaires, including the comparative version of the Speech, Spatial, and Qualities, the Listening Inventory for Education Revised, and the Children's Home Inventory for Listening Difficulties. Speech recognition in noise was typically better with EAS compared to participants' performance with acoustic- and electric-only stimulation, particularly when evaluated at the less favorable SNR. Additionally, in real-world situations, children generally preferred to use EAS compared to electric-only stimulation. Also, the participants' classroom teachers observed better hearing performance in the classroom with the use of EAS. Use of EAS provided better speech recognition in quiet and in noise when compared to performance obtained with use of acoustic- and electric-only stimulation, and children responded favorably to the use of EAS implemented in an integrated sound processor for real-world use. American Academy of Audiology

Neural Recruitment for the Production of Native and Novel Speech Sounds

PubMed Central

Moser, Dana; Fridriksson, Julius; Bonilha, Leonardo; Healy, Eric W.; Baylis, Gordon; Baker, Julie; Rorden, Chris

2010-01-01

Two primary areas of damage have been implicated in apraxia of speech (AOS) based on the time post-stroke: (1) the left inferior frontal gyrus (IFG) in acute patients, and (2) the left anterior insula (aIns) in chronic patients. While AOS is widely characterized as a disorder in motor speech planning, little is known about the specific contributions of each of these regions in speech. The purpose of this study was to investigate cortical activation during speech production with a specific focus on the aIns and the IFG in normal adults. While undergoing sparse fMRI, 30 normal adults completed a 30-minute speech-repetition task consisting of three-syllable nonwords that contained either (a) English (native) syllables or (b) Non-English (novel) syllables. When the novel syllable productions were compared to the native syllable productions, greater neural activation was observed in the aIns and IFG, particularly during the first 10 minutes of the task when novelty was the greatest. Although activation in the aIns remained high throughout the task for novel productions, greater activation was clearly demonstrated when the initial 10 minutes were compared to the final 10 minutes of the task. These results suggest increased activity within an extensive neural network, including the aIns and IFG, when the motor speech system is taxed, such as during the production of novel speech. We speculate that the amount of left aIns recruitment during speech production may be related to the internal construction of the motor speech unit such that the degree of novelty/automaticity would result in more or less demands respectively. The role of the IFG as a storehouse and integrative processor for previously acquired routines is also discussed. PMID:19385020

Development of flame resistant treatment for nomex fibrous structures

NASA Technical Reports Server (NTRS)

Toy, M. S.

1978-01-01

Technology which renders aramid fibrous structures flame resistant through chemical modification was developed. The project scaled up flame resistant treatment from laboratory fabric swatches of a few inches to efficiently producing ten yards of commercial width (41 inches) aromatic polyamide. The radiation intensity problem of the processor was resolved. Further improvement of the processor cooling system was recommended for two reasons: (1) To advance current technology of flame proofing Nomex fabric to higher oxygen enriched atmospheres; and (2) To adapt the processor for direct applicability to low cost commercial fabrics.

Modem design for a MOBILESAT terminal

NASA Technical Reports Server (NTRS)

Rice, M.; Miller, M. J.; Cowley, W. G.; Rowe, D.

1990-01-01

The implementation is described of a programmable digital signal processor based system, designed for use as a test bed in the development of a digital modem, codec, and channel simulator. Code was written to configure the system as a 5600 bps or 6600 bps QPSK modem. The test bed is currently being used in an experiment to evaluate the performance of digital speech over shadowed channels in the Australian mobile satellite (MOBILESAT) project.

Skin Necrosis After Implantation With the BAHA Attract: A Case Report and Review of the Literature.

PubMed

Chen, Stephanie Y; Mancuso, Dean; Lalwani, Anil K

2017-03-01

The bone-anchored hearing aid (BAHA) Attract is a transcutaneous bone conduction hearing aid that uses magnetic coupling to enable sound conduction. We report the first case of skin necrosis associated with the BAHA Attract and perform a literature review of soft tissue complications related to the device. A single patient who was found to develop skin necrosis 2 weeks after being fitted for the BAHA Attract speech processor. After the patient developed skin necrosis from the device, she was advised to immediately discontinue use of the Attract to allow complete wound healing, upon which the Attract was successfully converted to a percutaneous BAHA. We monitored for the development of skin complications from the BAHA Attract. The patient's immediate postoperative course was unremarkable and she was fitted with a speech processor of M5 magnet strength at 1 month postoperatively. After 1 week of use, she reported discomfort and was advised to downgrade to an M4 magnet; however, she continued to use the M5 and the following week was found to have developed skin necrosis around the device. Despite the infrequency of skin necrosis related to the BAHA Attract, it must be considered in counseling and managing candidates for the device.

A FPGA-Based, Granularity-Variable Neuromorphic Processor and Its Application in a MIMO Real-Time Control System.

PubMed

Zhang, Zhen; Ma, Cheng; Zhu, Rong

2017-08-23

Artificial Neural Networks (ANNs), including Deep Neural Networks (DNNs), have become the state-of-the-art methods in machine learning and achieved amazing success in speech recognition, visual object recognition, and many other domains. There are several hardware platforms for developing accelerated implementation of ANN models. Since Field Programmable Gate Array (FPGA) architectures are flexible and can provide high performance per watt of power consumption, they have drawn a number of applications from scientists. In this paper, we propose a FPGA-based, granularity-variable neuromorphic processor (FBGVNP). The traits of FBGVNP can be summarized as granularity variability, scalability, integrated computing, and addressing ability: first, the number of neurons is variable rather than constant in one core; second, the multi-core network scale can be extended in various forms; third, the neuron addressing and computing processes are executed simultaneously. These make the processor more flexible and better suited for different applications. Moreover, a neural network-based controller is mapped to FBGVNP and applied in a multi-input, multi-output, (MIMO) real-time, temperature-sensing and control system. Experiments validate the effectiveness of the neuromorphic processor. The FBGVNP provides a new scheme for building ANNs, which is flexible, highly energy-efficient, and can be applied in many areas.

A FPGA-Based, Granularity-Variable Neuromorphic Processor and Its Application in a MIMO Real-Time Control System

PubMed Central

Zhang, Zhen; Zhu, Rong

2017-01-01

Artificial Neural Networks (ANNs), including Deep Neural Networks (DNNs), have become the state-of-the-art methods in machine learning and achieved amazing success in speech recognition, visual object recognition, and many other domains. There are several hardware platforms for developing accelerated implementation of ANN models. Since Field Programmable Gate Array (FPGA) architectures are flexible and can provide high performance per watt of power consumption, they have drawn a number of applications from scientists. In this paper, we propose a FPGA-based, granularity-variable neuromorphic processor (FBGVNP). The traits of FBGVNP can be summarized as granularity variability, scalability, integrated computing, and addressing ability: first, the number of neurons is variable rather than constant in one core; second, the multi-core network scale can be extended in various forms; third, the neuron addressing and computing processes are executed simultaneously. These make the processor more flexible and better suited for different applications. Moreover, a neural network-based controller is mapped to FBGVNP and applied in a multi-input, multi-output, (MIMO) real-time, temperature-sensing and control system. Experiments validate the effectiveness of the neuromorphic processor. The FBGVNP provides a new scheme for building ANNs, which is flexible, highly energy-efficient, and can be applied in many areas. PMID:28832522

Technology and Speech Training: An Affair to Remember.

ERIC Educational Resources Information Center

Levitt, Harry

1989-01-01

A history of speech training technology is presented, from the simple hand-held mirror to complicated computer-based systems and tactile devices, and subsequent papers in this theme issue are introduced. Both the advantages and problems of technological aids are addressed. Simplicity in the application and use of speech training aids is stressed.…

Computer Sciences and Data Systems, volume 2

NASA Technical Reports Server (NTRS)

1987-01-01

Topics addressed include: data storage; information network architecture; VHSIC technology; fiber optics; laser applications; distributed processing; spaceborne optical disk controller; massively parallel processors; and advanced digital SAR processors.

Development Status of the International Space Station Urine Processor Assembly

NASA Technical Reports Server (NTRS)

Holder, Donald W.; Hutchens, Cindy F.

2003-01-01

NASA, Marshall Space Flight Center (MSFC) is developing a Urine Processor Assembly (UPA) for the International Space Station (ISS). The UPA uses Vapor Compression Distillation (VCD) technology to reclaim water from pre-treated urine. This water is further processed by the Water Processor Assembly (WPA) to potable quality standards for use on the ISS. NASA has developed this technology over the last 25-30 years. Over this history, many technical issues were solved with thousands of hours of ground testing that demonstrate the ability of the UPA technology to reclaim water from urine. In recent years, NASA MSFC has been responsible for taking the UPA technology to "flight design" maturity. This paper will give a brief overview of the UPA design and a status of the major design and development efforts completed recently to mature the UPA to a flight level.

Ion Thruster Development at NASA Lewis Research Center

NASA Technical Reports Server (NTRS)

Sovey, James S.; Hamley, John A.; Patterson, Michael J.; Rawlin, Vincent K.; Sarver-Verhey, Timothy R.

1992-01-01

Recent ion propulsion technology efforts at NASA's Lewis Research Center including development of kW-class xenon ion thrusters, high power xenon and krypton ion thrusters, and power processors are reviewed. Thruster physical characteristics, performance data, life projections, and power processor component technology are summarized. The ion propulsion technology program is structured to address a broad set of mission applications from satellite stationkeeping and repositioning to primary propulsion using solar or nuclear power systems.

78 FR 70888 - Need for Agency Approval of a Railroad's Use of Certain Technology That Has Been Previously...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-11-27

... technologies, namely safety-critical processor-based signal or train control systems, including subsystems and... or train control system (including a subsystem or component thereof) that was in service as of June 6... processor-based signal or train control system, subsystem, or component.'' See 49 CFR 236.903. Under Subpart...

Radiation Hardened Electronics for Extreme Environments

NASA Technical Reports Server (NTRS)

Keys, Andrew S.; Watson, Michael D.

2007-01-01

The Radiation Hardened Electronics for Space Environments (RHESE) project consists of a series of tasks designed to develop and mature a broad spectrum of radiation hardened and low temperature electronics technologies. Three approaches are being taken to address radiation hardening: improved material hardness, design techniques to improve radiation tolerance, and software methods to improve radiation tolerance. Within these approaches various technology products are being addressed including Field Programmable Gate Arrays (FPGA), Field Programmable Analog Arrays (FPAA), MEMS Serial Processors, Reconfigurable Processors, and Parallel Processors. In addition to radiation hardening, low temperature extremes are addressed with a focus on material and design approaches.

Digital signal processing algorithms for automatic voice recognition

NASA Technical Reports Server (NTRS)

Botros, Nazeih M.

1987-01-01

The current digital signal analysis algorithms are investigated that are implemented in automatic voice recognition algorithms. Automatic voice recognition means, the capability of a computer to recognize and interact with verbal commands. The digital signal is focused on, rather than the linguistic, analysis of speech signal. Several digital signal processing algorithms are available for voice recognition. Some of these algorithms are: Linear Predictive Coding (LPC), Short-time Fourier Analysis, and Cepstrum Analysis. Among these algorithms, the LPC is the most widely used. This algorithm has short execution time and do not require large memory storage. However, it has several limitations due to the assumptions used to develop it. The other 2 algorithms are frequency domain algorithms with not many assumptions, but they are not widely implemented or investigated. However, with the recent advances in the digital technology, namely signal processors, these 2 frequency domain algorithms may be investigated in order to implement them in voice recognition. This research is concerned with real time, microprocessor based recognition algorithms.

Wireless Source Localization and Signal Collection from an Airborne Symmetric Line Array Sensor Network

DTIC Science & Technology

2014-09-01

band signal samples by taking the ratio of (166) and (165) as     2 2 /2 /2 sin sin coscos g g g g gg cQ cI eE n E n e...processors,” EEE Trans. Acoust. Speech Signal Process., vol. 31, no. 6, pp. 1378–1393, Dec. 1983. [10] J. Li, P. Stoica and Z. Wang, “On robust

Evaluation of Speech Recognition of Cochlear Implant Recipients Using Adaptive, Digital Remote Microphone Technology and a Speech Enhancement Sound Processing Algorithm.

PubMed

Wolfe, Jace; Morais, Mila; Schafer, Erin; Agrawal, Smita; Koch, Dawn

2015-05-01

Cochlear implant recipients often experience difficulty with understanding speech in the presence of noise. Cochlear implant manufacturers have developed sound processing algorithms designed to improve speech recognition in noise, and research has shown these technologies to be effective. Remote microphone technology utilizing adaptive, digital wireless radio transmission has also been shown to provide significant improvement in speech recognition in noise. There are no studies examining the potential improvement in speech recognition in noise when these two technologies are used simultaneously. The goal of this study was to evaluate the potential benefits and limitations associated with the simultaneous use of a sound processing algorithm designed to improve performance in noise (Advanced Bionics ClearVoice) and a remote microphone system that incorporates adaptive, digital wireless radio transmission (Phonak Roger). A two-by-two way repeated measures design was used to examine performance differences obtained without these technologies compared to the use of each technology separately as well as the simultaneous use of both technologies. Eleven Advanced Bionics (AB) cochlear implant recipients, ages 11 to 68 yr. AzBio sentence recognition was measured in quiet and in the presence of classroom noise ranging in level from 50 to 80 dBA in 5-dB steps. Performance was evaluated in four conditions: (1) No ClearVoice and no Roger, (2) ClearVoice enabled without the use of Roger, (3) ClearVoice disabled with Roger enabled, and (4) simultaneous use of ClearVoice and Roger. Speech recognition in quiet was better than speech recognition in noise for all conditions. Use of ClearVoice and Roger each provided significant improvement in speech recognition in noise. The best performance in noise was obtained with the simultaneous use of ClearVoice and Roger. ClearVoice and Roger technology each improves speech recognition in noise, particularly when used at the same time. Because ClearVoice does not degrade performance in quiet settings, clinicians should consider recommending ClearVoice for routine, full-time use for AB implant recipients. Roger should be used in all instances in which remote microphone technology may assist the user in understanding speech in the presence of noise. American Academy of Audiology.

Processing F0 with cochlear implants: Modulation frequency discrimination and speech intonation recognition.

PubMed

Chatterjee, Monita; Peng, Shu-Chen

2008-01-01

Fundamental frequency (F0) processing by cochlear implant (CI) listeners was measured using a psychophysical task and a speech intonation recognition task. Listeners' Weber fractions for modulation frequency discrimination were measured using an adaptive, 3-interval, forced-choice paradigm: stimuli were presented through a custom research interface. In the speech intonation recognition task, listeners were asked to indicate whether resynthesized bisyllabic words, when presented in the free field through the listeners' everyday speech processor, were question-like or statement-like. The resynthesized tokens were systematically manipulated to have different initial-F0s to represent male vs. female voices, and different F0 contours (i.e. falling, flat, and rising) Although the CI listeners showed considerable variation in performance on both tasks, significant correlations were observed between the CI listeners' sensitivity to modulation frequency in the psychophysical task and their performance in intonation recognition. Consistent with their greater reliance on temporal cues, the CI listeners' performance in the intonation recognition task was significantly poorer with the higher initial-F0 stimuli than with the lower initial-F0 stimuli. Similar results were obtained with normal hearing listeners attending to noiseband-vocoded CI simulations with reduced spectral resolution.

Processing F0 with Cochlear Implants: Modulation Frequency Discrimination and Speech Intonation Recognition

PubMed Central

Chatterjee, Monita; Peng, Shu-Chen

2008-01-01

Fundamental frequency (F0) processing by cochlear implant (CI) listeners was measured using a psychophysical task and a speech intonation recognition task. Listeners’ Weber fractions for modulation frequency discrimination were measured using an adaptive, 3-interval, forced-choice paradigm: stimuli were presented through a custom research interface. In the speech intonation recognition task, listeners were asked to indicate whether resynthesized bisyllabic words, when presented in the free field through the listeners’ everyday speech processor, were question-like or statement-like. The resynthesized tokens were systematically manipulated to have different initial F0s to represent male vs. female voices, and different F0 contours (i.e., falling, flat, and rising) Although the CI listeners showed considerable variation in performance on both tasks, significant correlations were observed between the CI listeners’ sensitivity to modulation frequency in the psychophysical task and their performance in intonation recognition. Consistent with their greater reliance on temporal cues, the CI listeners’ performance in the intonation recognition task was significantly poorer with the higher initial-F0 stimuli than with the lower initial-F0 stimuli. Similar results were obtained with normal hearing listeners attending to noiseband-vocoded CI simulations with reduced spectral resolution. PMID:18093766

Effect of input compression and input frequency response on music perception in cochlear implant users.

PubMed

Halliwell, Emily R; Jones, Linor L; Fraser, Matthew; Lockley, Morag; Hill-Feltham, Penelope; McKay, Colette M

2015-06-01

A study was conducted to determine whether modifications to input compression and input frequency response characteristics can improve music-listening satisfaction in cochlear implant users. Experiment 1 compared three pre-processed versions of music and speech stimuli in a laboratory setting: original, compressed, and flattened frequency response. Music excerpts comprised three music genres (classical, country, and jazz), and a running speech excerpt was compared. Experiment 2 implemented a flattened input frequency response in the speech processor program. In a take-home trial, participants compared unaltered and flattened frequency responses. Ten and twelve adult Nucleus Freedom cochlear implant users participated in Experiments 1 and 2, respectively. Experiment 1 revealed a significant preference for music stimuli with a flattened frequency response compared to both original and compressed stimuli, whereas there was a significant preference for the original (rising) frequency response for speech stimuli. Experiment 2 revealed no significant mean preference for the flattened frequency response, with 9 of 11 subjects preferring the rising frequency response. Input compression did not alter music enjoyment. Comparison of the two experiments indicated that individual frequency response preferences may depend on the genre or familiarity, and particularly whether the music contained lyrics.

Coding strategies for cochlear implants under adverse environments

NASA Astrophysics Data System (ADS)

Tahmina, Qudsia

Cochlear implants are electronic prosthetic devices that restores partial hearing in patients with severe to profound hearing loss. Although most coding strategies have significantly improved the perception of speech in quite listening conditions, there remains limitations on speech perception under adverse environments such as in background noise, reverberation and band-limited channels, and we propose strategies that improve the intelligibility of speech transmitted over the telephone networks, reverberated speech and speech in the presence of background noise. For telephone processed speech, we propose to examine the effects of adding low-frequency and high- frequency information to the band-limited telephone speech. Four listening conditions were designed to simulate the receiving frequency characteristics of telephone handsets. Results indicated improvement in cochlear implant and bimodal listening when telephone speech was augmented with high frequency information and therefore this study provides support for design of algorithms to extend the bandwidth towards higher frequencies. The results also indicated added benefit from hearing aids for bimodal listeners in all four types of listening conditions. Speech understanding in acoustically reverberant environments is always a difficult task for hearing impaired listeners. Reverberated sounds consists of direct sound, early reflections and late reflections. Late reflections are known to be detrimental to speech intelligibility. In this study, we propose a reverberation suppression strategy based on spectral subtraction to suppress the reverberant energies from late reflections. Results from listening tests for two reverberant conditions (RT60 = 0.3s and 1.0s) indicated significant improvement when stimuli was processed with SS strategy. The proposed strategy operates with little to no prior information on the signal and the room characteristics and therefore, can potentially be implemented in real-time CI speech processors. For speech in background noise, we propose a mechanism underlying the contribution of harmonics to the benefit of electroacoustic stimulations in cochlear implants. The proposed strategy is based on harmonic modeling and uses synthesis driven approach to synthesize the harmonics in voiced segments of speech. Based on objective measures, results indicated improvement in speech quality. This study warrants further work into development of algorithms to regenerate harmonics of voiced segments in the presence of noise.

Pre- and Postoperative Binaural Unmasking for Bimodal Cochlear Implant Listeners.

PubMed

Sheffield, Benjamin M; Schuchman, Gerald; Bernstein, Joshua G W

Cochlear implants (CIs) are increasingly recommended to individuals with residual bilateral acoustic hearing. Although new hearing-preserving electrode designs and surgical approaches show great promise, CI recipients are still at risk to lose acoustic hearing in the implanted ear, which could prevent the ability to take advantage of binaural unmasking to aid speech recognition in noise. This study examined the tradeoff between the benefits of a CI for speech understanding in noise and the potential loss of binaural unmasking for CI recipients with some bilateral preoperative acoustic hearing. Binaural unmasking is difficult to evaluate in CI candidates because speech perception in noise is generally too poor to measure reliably in the range of signal to noise ratios (SNRs) where binaural intelligibility level differences (BILDs) are typically observed (<5 dB). Thus, a test of audiovisual speech perception in noise was employed to increase performance to measureable levels. BILDs were measured preoperatively for 11 CI candidates and at least 5 months post-activation for 10 of these individuals (1 individual elected not to receive a CI). Audiovisual sentences were presented in speech-shaped masking noise between -10 and +15 dB SNR. The noise was always correlated between the ears, while the speech signal was either correlated (N0S0) or inversely correlated (N0Sπ). Stimuli were delivered via headphones to the unaided ear(s) and, where applicable, via auxiliary input to the CI speech processor. A z test evaluated performance differences between the N0S0 and N0Sπ conditions for each listener pre- and postoperatively. For listeners showing a significant difference, the magnitude of the BILD was characterized as the difference in SNRs required to achieve 50% correct performance. One listener who underwent hearing-preservation surgery received additional postoperative tests, which presented sound directly to both ears and to the CI speech processor. Five of 11 listeners showed a significant preoperative BILD (range: 2.0 to 7.3 dB). Only 2 of these 5 showed a significant postoperative BILD, but the mean BILD was smaller (1.3 dB) than that observed preoperatively (3.1 dB). Despite the fact that some listeners lost the preoperative binaural benefit, 9 out of 10 listeners tested postoperatively had performance equal to or better than their best pre-CI performance. The listener who retained functional acoustic hearing in the implanted ear also demonstrated a preserved acoustic BILD postoperatively. Approximately half of the CI candidates in this study demonstrated preoperative binaural hearing benefits for audiovisual speech perception in noise. Most of these listeners lost their acoustic hearing in the implanted ear after surgery (using nonhearing-preservation techniques), and therefore lost access to this binaural benefit. In all but one case, any loss of binaural benefit was compensated for or exceeded by an improvement in speech perception with the CI. Evidence of a preoperative BILD suggests that certain CI candidates might further benefit from hearing-preservation surgery to retain acoustic binaural unmasking, as demonstrated for the listener who underwent hearing-preservation surgery. This test of binaural audiovisual speech perception in noise could serve as a diagnostic tool to identify CI candidates who are most likely to receive functional benefits from their bilateral acoustic hearing.

Spectro-temporal cues enhance modulation sensitivity in cochlear implant users

PubMed Central

Zheng, Yi; Escabí, Monty; Litovsky, Ruth Y.

2018-01-01

Although speech understanding is highly variable amongst cochlear implants (CIs) subjects, the remarkably high speech recognition performance of many CI users is unexpected and not well understood. Numerous factors, including neural health and degradation of the spectral information in the speech signal of CIs, likely contribute to speech understanding. We studied the ability to use spectro-temporal modulations, which may be critical for speech understanding and discrimination, and hypothesize that CI users adopt a different perceptual strategy than normal-hearing (NH) individuals, whereby they rely more heavily on joint spectro-temporal cues to enhance detection of auditory cues. Modulation detection sensitivity was studied in CI users and NH subjects using broadband “ripple” stimuli that were modulated spectrally, temporally, or jointly, i.e., spectro-temporally. The spectro-temporal modulation transfer functions of CI users and NH subjects was decomposed into spectral and temporal dimensions and compared to those subjects’ spectral-only and temporal-only modulation transfer functions. In CI users, the joint spectro-temporal sensitivity was better than that predicted by spectral-only and temporal-only sensitivity, indicating a heightened spectro-temporal sensitivity. Such an enhancement through the combined integration of spectral and temporal cues was not observed in NH subjects. The unique use of spectro-temporal cues by CI patients can yield benefits for use of cues that are important for speech understanding. This finding has implications for developing sound processing strategies that may rely on joint spectro-temporal modulations to improve speech comprehension of CI users, and the findings of this study may be valuable for developing clinical assessment tools to optimize CI processor performance. PMID:28601530

Automatic Speech Acquisition and Recognition for Spacesuit Audio Systems

NASA Technical Reports Server (NTRS)

Ye, Sherry

2015-01-01

NASA has a widely recognized but unmet need for novel human-machine interface technologies that can facilitate communication during astronaut extravehicular activities (EVAs), when loud noises and strong reverberations inside spacesuits make communication challenging. WeVoice, Inc., has developed a multichannel signal-processing method for speech acquisition in noisy and reverberant environments that enables automatic speech recognition (ASR) technology inside spacesuits. The technology reduces noise by exploiting differences between the statistical nature of signals (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, ASR accuracy can be improved to the level at which crewmembers will find the speech interface useful. System components and features include beam forming/multichannel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, and ASR decoding. Arithmetic complexity models were developed and will help designers of real-time ASR systems select proper tasks when confronted with constraints in computational resources. In Phase I of the project, WeVoice validated the technology. The company further refined the technology in Phase II and developed a prototype for testing and use by suited astronauts.

The Impact of 3D Stacking and Technology Scaling on the Power and Area of Stereo Matching Processors.

PubMed

Ok, Seung-Ho; Lee, Yong-Hwan; Shim, Jae Hoon; Lim, Sung Kyu; Moon, Byungin

2017-02-22

Recently, stereo matching processors have been adopted in real-time embedded systems such as intelligent robots and autonomous vehicles, which require minimal hardware resources and low power consumption. Meanwhile, thanks to the through-silicon via (TSV), three-dimensional (3D) stacking technology has emerged as a practical solution to achieving the desired requirements of a high-performance circuit. In this paper, we present the benefits of 3D stacking and process technology scaling on stereo matching processors. We implemented 2-tier 3D-stacked stereo matching processors with GlobalFoundries 130-nm and Nangate 45-nm process design kits and compare them with their two-dimensional (2D) counterparts to identify comprehensive design benefits. In addition, we examine the findings from various analyses to identify the power benefits of 3D-stacked integrated circuit (IC) and device technology advancements. From experiments, we observe that the proposed 3D-stacked ICs, compared to their 2D IC counterparts, obtain 43% area, 13% power, and 14% wire length reductions. In addition, we present a logic partitioning method suitable for a pipeline-based hardware architecture that minimizes the use of TSVs.

The Impact of 3D Stacking and Technology Scaling on the Power and Area of Stereo Matching Processors

PubMed Central

Ok, Seung-Ho; Lee, Yong-Hwan; Shim, Jae Hoon; Lim, Sung Kyu; Moon, Byungin

2017-01-01

Recently, stereo matching processors have been adopted in real-time embedded systems such as intelligent robots and autonomous vehicles, which require minimal hardware resources and low power consumption. Meanwhile, thanks to the through-silicon via (TSV), three-dimensional (3D) stacking technology has emerged as a practical solution to achieving the desired requirements of a high-performance circuit. In this paper, we present the benefits of 3D stacking and process technology scaling on stereo matching processors. We implemented 2-tier 3D-stacked stereo matching processors with GlobalFoundries 130-nm and Nangate 45-nm process design kits and compare them with their two-dimensional (2D) counterparts to identify comprehensive design benefits. In addition, we examine the findings from various analyses to identify the power benefits of 3D-stacked integrated circuit (IC) and device technology advancements. From experiments, we observe that the proposed 3D-stacked ICs, compared to their 2D IC counterparts, obtain 43% area, 13% power, and 14% wire length reductions. In addition, we present a logic partitioning method suitable for a pipeline-based hardware architecture that minimizes the use of TSVs. PMID:28241437

Design and Development of a Baseband Processor for the Advanced Communications Technology Satellite

NASA Technical Reports Server (NTRS)

Lee, Kerry D.

1996-01-01

This paper describes the implementation of the operational baseband processor (BBP) subsystem on board the NASA Advanced Communications Technology Satellite (ACTS). The BBP supports the network consisting of the NASA ground station (NGS) low burst rate (LBR) terminals, and the T1 very small aperture terminals (VSAT's), to provide flexible, demand assigned satellite switched (SS), baseband processed frequency division modulated (FDM)/time division multiple access (TDMA) operations. This paper presents an overview of the baseband processor and includes a description of the data flow, functional block diagrams, and a discussion of the implementation of BBP. A discussion of the supporting technologies for the BBP is presented. A brief summary of BBP-level performance testing is also presented. Finally, a discussion of the implications of current technology on the BBP design, if it were to be developed today, is presented.

DOE Office of Scientific and Technical Information (OSTI.GOV)

Reed, D.A.; Grunwald, D.C.

The spectrum of parallel processor designs can be divided into three sections according to the number and complexity of the processors. At one end there are simple, bit-serial processors. Any one of thee processors is of little value, but when it is coupled with many others, the aggregate computing power can be large. This approach to parallel processing can be likened to a colony of termites devouring a log. The most notable examples of this approach are the NASA/Goodyear Massively Parallel Processor, which has 16K one-bit processors, and the Thinking Machines Connection Machine, which has 64K one-bit processors. At themore » other end of the spectrum, a small number of processors, each built using the fastest available technology and the most sophisticated architecture, are combined. An example of this approach is the Cray X-MP. This type of parallel processing is akin to four woodmen attacking the log with chainsaws.« less

Graphics Processor Units (GPUs)

NASA Technical Reports Server (NTRS)

Wyrwas, Edward J.

2017-01-01

This presentation will include information about Graphics Processor Units (GPUs) technology, NASA Electronic Parts and Packaging (NEPP) tasks, The test setup, test parameter considerations, lessons learned, collaborations, a roadmap, NEPP partners, results to date, and future plans.

Practical applications of interactive voice technologies: Some accomplishments and prospects

NASA Technical Reports Server (NTRS)

Grady, Michael W.; Hicklin, M. B.; Porter, J. E.

1977-01-01

A technology assessment of the application of computers and electronics to complex systems is presented. Three existing systems which utilize voice technology (speech recognition and speech generation) are described. Future directions in voice technology are also described.

Voice-processing technologies--their application in telecommunications.

PubMed Central

Wilpon, J G

1995-01-01

As the telecommunications industry evolves over the next decade to provide the products and services that people will desire, several key technologies will become commonplace. Two of these, automatic speech recognition and text-to-speech synthesis, will provide users with more freedom on when, where, and how they access information. While these technologies are currently in their infancy, their capabilities are rapidly increasing and their deployment in today's telephone network is expanding. The economic impact of just one application, the automation of operator services, is well over $100 million per year. Yet there still are many technical challenges that must be resolved before these technologies can be deployed ubiquitously in products and services throughout the worldwide telephone network. These challenges include: (i) High level of accuracy. The technology must be perceived by the user as highly accurate, robust, and reliable. (ii) Easy to use. Speech is only one of several possible input/output modalities for conveying information between a human and a machine, much like a computer terminal or Touch-Tone pad on a telephone. It is not the final product. Therefore, speech technologies must be hidden from the user. That is, the burden of using the technology must be on the technology itself. (iii) Quick prototyping and development of new products and services. The technology must support the creation of new products and services based on speech in an efficient and timely fashion. In this paper I present a vision of the voice-processing industry with a focus on the areas with the broadest base of user penetration: speech recognition, text-to-speech synthesis, natural language processing, and speaker recognition technologies. The current and future applications of these technologies in the telecommunications industry will be examined in terms of their strengths, limitations, and the degree to which user needs have been or have yet to be met. Although noteworthy gains have been made in areas with potentially small user bases and in the more mature speech-coding technologies, these subjects are outside the scope of this paper. Images Fig. 1 PMID:7479815

Instruction-level performance modeling and characterization of multimedia applications

DOE Office of Scientific and Technical Information (OSTI.GOV)

Luo, Y.; Cameron, K.W.

1999-06-01

One of the challenges for characterizing and modeling realistic multimedia applications is the lack of access to source codes. On-chip performance counters effectively resolve this problem by monitoring run-time behaviors at the instruction-level. This paper presents a novel technique of characterizing and modeling workloads at the instruction level for realistic multimedia applications using hardware performance counters. A variety of instruction counts are collected from some multimedia applications, such as RealPlayer, GSM Vocoder, MPEG encoder/decoder, and speech synthesizer. These instruction counts can be used to form a set of abstract characteristic parameters directly related to a processor`s architectural features. Based onmore » microprocessor architectural constraints and these calculated abstract parameters, the architectural performance bottleneck for a specific application can be estimated. Meanwhile, the bottleneck estimation can provide suggestions about viable architectural/functional improvement for certain workloads. The biggest advantage of this new characterization technique is a better understanding of processor utilization efficiency and architectural bottleneck for each application. This technique also provides predictive insight of future architectural enhancements and their affect on current codes. In this paper the authors also attempt to model architectural effect on processor utilization without memory influence. They derive formulas for calculating CPI{sub 0}, CPI without memory effect, and they quantify utilization of architectural parameters. These equations are architecturally diagnostic and predictive in nature. Results provide promise in code characterization, and empirical/analytical modeling.« less

Objective speech transmission improvements with a binaural cochlear implant sound-coding strategy inspired by the contralateral medial olivocochlear reflex.

PubMed

Lopez-Poveda, Enrique A; Eustaquio-Martín, Almudena

2018-04-01

It has been recently shown that cochlear implant users could enjoy better speech reception in noise and enhanced spatial unmasking with binaural audio processing inspired by the inhibitory effects of the contralateral medial olivocochlear (MOC) reflex on compression [Lopez-Poveda, Eustaquio-Martin, Stohl, Wolford, Schatzer, and Wilson (2016). Ear Hear. 37, e138-e148]. The perceptual evidence supporting those benefits, however, is limited to a few target-interferer spatial configurations and to a particular implementation of contralateral MOC inhibition. Here, the short-term objective intelligibility index is used to (1) objectively demonstrate potential benefits over many more spatial configurations, and (2) investigate if the predicted benefits may be enhanced by using more realistic MOC implementations. Results corroborate the advantages and drawbacks of MOC processing indicated by the previously published perceptual tests. The results also suggest that the benefits may be enhanced and the drawbacks overcome by using longer time constants for the activation and deactivation of inhibition and, to a lesser extent, by using a comparatively greater inhibition in the lower than in the higher frequency channels. Compared to using two functionally independent processors, the better MOC processor improved the signal-to-noise ratio in the two ears between 1 and 6 decibels by enhancing head-shadow effects, and was advantageous for all tested target-interferer spatial configurations.

Speech perception for adult cochlear implant recipients in a realistic background noise: effectiveness of preprocessing strategies and external options for improving speech recognition in noise.

PubMed

Gifford, René H; Revit, Lawrence J

2010-01-01

Although cochlear implant patients are achieving increasingly higher levels of performance, speech perception in noise continues to be problematic. The newest generations of implant speech processors are equipped with preprocessing and/or external accessories that are purported to improve listening in noise. Most speech perception measures in the clinical setting, however, do not provide a close approximation to real-world listening environments. To assess speech perception for adult cochlear implant recipients in the presence of a realistic restaurant simulation generated by an eight-loudspeaker (R-SPACE) array in order to determine whether commercially available preprocessing strategies and/or external accessories yield improved sentence recognition in noise. Single-subject, repeated-measures design with two groups of participants: Advanced Bionics and Cochlear Corporation recipients. Thirty-four subjects, ranging in age from 18 to 90 yr (mean 54.5 yr), participated in this prospective study. Fourteen subjects were Advanced Bionics recipients, and 20 subjects were Cochlear Corporation recipients. Speech reception thresholds (SRTs) in semidiffuse restaurant noise originating from an eight-loudspeaker array were assessed with the subjects' preferred listening programs as well as with the addition of either Beam preprocessing (Cochlear Corporation) or the T-Mic accessory option (Advanced Bionics). In Experiment 1, adaptive SRTs with the Hearing in Noise Test sentences were obtained for all 34 subjects. For Cochlear Corporation recipients, SRTs were obtained with their preferred everyday listening program as well as with the addition of Focus preprocessing. For Advanced Bionics recipients, SRTs were obtained with the integrated behind-the-ear (BTE) mic as well as with the T-Mic. Statistical analysis using a repeated-measures analysis of variance (ANOVA) evaluated the effects of the preprocessing strategy or external accessory in reducing the SRT in noise. In addition, a standard t-test was run to evaluate effectiveness across manufacturer for improving the SRT in noise. In Experiment 2, 16 of the 20 Cochlear Corporation subjects were reassessed obtaining an SRT in noise using the manufacturer-suggested "Everyday," "Noise," and "Focus" preprocessing strategies. A repeated-measures ANOVA was employed to assess the effects of preprocessing. The primary findings were (i) both Noise and Focus preprocessing strategies (Cochlear Corporation) significantly improved the SRT in noise as compared to Everyday preprocessing, (ii) the T-Mic accessory option (Advanced Bionics) significantly improved the SRT as compared to the BTE mic, and (iii) Focus preprocessing and the T-Mic resulted in similar degrees of improvement that were not found to be significantly different from one another. Options available in current cochlear implant sound processors are able to significantly improve speech understanding in a realistic, semidiffuse noise with both Cochlear Corporation and Advanced Bionics systems. For Cochlear Corporation recipients, Focus preprocessing yields the best speech-recognition performance in a complex listening environment; however, it is recommended that Noise preprocessing be used as the new default for everyday listening environments to avoid the need for switching programs throughout the day. For Advanced Bionics recipients, the T-Mic offers significantly improved performance in noise and is recommended for everyday use in all listening environments. American Academy of Audiology.

Hearing Preservation Outcomes After Cochlear Implantation Depending on the Angle of Insertion: Indication for Electric or Electric-Acoustic Stimulation.

PubMed

Helbig, Silke; Adel, Youssef; Leinung, Martin; Stöver, Timo; Baumann, Uwe; Weissgerber, Tobias

2018-06-15

This study reviewed outcomes of hearing preservation (HP) surgery depending on the angle of insertion (AOI) in a cochlear implant (CI) patient population who used electric stimulation (ES) or combined electric-acoustic stimulation (EAS). Retrospective case review. Tertiary referral university hospital. Ninety-one patients with different degrees of preoperative low-frequency residual hearing who underwent HP surgery with a free-fitting lateral-wall electrode array (MED-EL Flex) with lengths ranging from 20.0 to 31.5 mm. Cochlear implantation using HP surgery technique and subsequent fitting with CI speech processor for ES, or combined CI and hearing aid speech processor for EAS. Individual AOI were estimated using modified Stenvers' projection. Freiburg monosyllable test in quiet (free-field presentation at 65 dB SPL) and pure-tone averages for low frequencies (125, 250, and 500 Hz; PTAlow) were evaluated during a follow-up period of 12 months after implantation. Estimated AOIs showed bimodal distribution: shallow insertion (SI) with mean AOI of 377 degrees and deep insertion (DI) with mean AOI of 608 degrees. Speech test scores after 12 months were comparable between AOI groups, however, they were significantly different between stimulation types with better scores for EAS. Only ES showed a positive correlation (r = 0.293) between speech test score and AOI. When HP was possible, both SI and DI showed significant postoperative PTAlow shifts with mean of 17.8 and 21.6 dB, respectively. These were comparable between AOI groups and no significant shifts were observed in follow-up intervals. Audiometric indication for HP and subsequent EAS is proposed up to 65 dB HL at 500 Hz, and up to 87 dB HL for HP. CI candidates can benefit from HP surgery with deep insertion when only using ES due to insufficient residual hearing. Conversely, candidates with preoperative threshold up to 65 dB HL at 500 Hz could perform significantly better with EAS which requires shallow insertion.

Bone anchored hearing aid: an evidence-based analysis.

PubMed

2002-01-01

The objective of this health technology policy assessment was to determine the effectiveness and cost-effectiveness of bone-anchored hearing aid (BAHA) in improving the hearing of people with conduction or mixed hearing loss. The (BAHA) is a bone conduction hearing device that includes a titanium fixture permanently implanted into the mastoid bone of the skull and an external percutaneous sound processor. The sound processor is attached to the fixture by means of a skin penetrating abutment. Because the device bypasses the middle ear and directly stimulates the cochlea, it has been recommended for individuals with conduction hearing loss or discharging middle ear infection. The titanium implant is expected to last a lifetime while the external sound processor is expected to last 5 years. The total initial device cost is approximately $5,300 and the external sound processor costs approximately $3,500. REVIEW OF BAHA BY THE MEDICAL ADVISORY SECRETARIAT: The Medical Advisory Secretariat's review is a descriptive synthesis of findings from 36 research articles published between January 1990 and May 2002. No randomized controlled studies were found. The evidence was derived from level 4 case series with relative small sample sizes (ranging from 30-188). The majority of the studies have follow-up periods of eight years or longer. All except one study were based on monaural BAHA implant on the side with the best bone conduction threshold. Level 4 evidence showed that BAHA has been be implanted safely in adults and children with success rates of 90% or higher in most studies. No mortality or life threatening morbidity has been reported. Revision rates for tissue reduction or resiting were generally under 10% for adults but have been reported to be as high as 25% in pediatric studies. Adverse skin reaction around the skin penetration site was the most common complication reported. Most of these conditions were successfully treated with antibiotics, and only 1% to 2% required surgical revision. Less than 1% required removal of the fixture. Other complications included failure to osseointegrate and loss of fixture and/or abutment due to trauma or infection. Studies showed that BAHAs were implanted in people who have conduction or mixed hearing loss, congenital atresia or suppurative otitis media who were not candidates for surgical repair, and who cannot use conventional bone conduction hearing aids. The need for BAHA is not age- related. Objective audiometric measures and subjective patient satisfaction surveys showed that BAHA significantly improved the unaided and aided free field and sound field thresholds as well as speech discrimination in quiet and in noise for former users of conventional bone conduction hearing aids. The outcomes were ambiguous for former users of air conduction hearing aids. BAHA has been shown to reduce the frequency of ear infection and reduce the discharge particularly among patients with suppurative otitis media. Patients have reported that BAHA improved their quality of life. Reported benefits were improved speech intelligibility, better sound comfort, less pressure on the head, less skin irritation, greater cosmetic acceptance and increase in confidence. Main reported shortcomings were wind noise, feedback and difficulty in using the telephone. Experts and the BAHA manufacturer recommended that recipients of a BAHA implant be at least 5 years old. Challenges associated with the implantation of BAHA in pediatric patients include thin bone, soft bone, higher rates of fixture loss due to trauma, psychological problems, and higher revision rates due to rapid bone growth. The overall outcomes are comparable to adult BAHA. The benefits of pediatric BAHA (e.g. on speech development) appear to outweigh the disadvantages. Screening according to strict eligibility criteria, preoperative counselling, close monitoring by a physician with BAHA expertise and on-going follow-up were identified as critical factors for long-term implant survival. Examples of eligibility criteria were provided. No literature on cost-effectiveness of BAHA was found.

Building Searchable Collections of Enterprise Speech Data.

ERIC Educational Resources Information Center

Cooper, James W.; Viswanathan, Mahesh; Byron, Donna; Chan, Margaret

The study has applied speech recognition and text-mining technologies to a set of recorded outbound marketing calls and analyzed the results. Since speaker-independent speech recognition technology results in a significantly lower recognition rate than that found when the recognizer is trained for a particular speaker, a number of post-processing…

System balance analysis for vector computers

NASA Technical Reports Server (NTRS)

Knight, J. C.; Poole, W. G., Jr.; Voight, R. G.

1975-01-01

The availability of vector processors capable of sustaining computing rates of 10 to the 8th power arithmetic results pers second raised the question of whether peripheral storage devices representing current technology can keep such processors supplied with data. By examining the solution of a large banded linear system on these computers, it was found that even under ideal conditions, the processors will frequently be waiting for problem data.

Technological evaluation of gesture and speech interfaces for enabling dismounted soldier-robot dialogue

NASA Astrophysics Data System (ADS)

Kattoju, Ravi Kiran; Barber, Daniel J.; Abich, Julian; Harris, Jonathan

2016-05-01

With increasing necessity for intuitive Soldier-robot communication in military operations and advancements in interactive technologies, autonomous robots have transitioned from assistance tools to functional and operational teammates able to service an array of military operations. Despite improvements in gesture and speech recognition technologies, their effectiveness in supporting Soldier-robot communication is still uncertain. The purpose of the present study was to evaluate the performance of gesture and speech interface technologies to facilitate Soldier-robot communication during a spatial-navigation task with an autonomous robot. Gesture and speech semantically based spatial-navigation commands leveraged existing lexicons for visual and verbal communication from the U.S Army field manual for visual signaling and a previously established Squad Level Vocabulary (SLV). Speech commands were recorded by a Lapel microphone and Microsoft Kinect, and classified by commercial off-the-shelf automatic speech recognition (ASR) software. Visual signals were captured and classified using a custom wireless gesture glove and software. Participants in the experiment commanded a robot to complete a simulated ISR mission in a scaled down urban scenario by delivering a sequence of gesture and speech commands, both individually and simultaneously, to the robot. Performance and reliability of gesture and speech hardware interfaces and recognition tools were analyzed and reported. Analysis of experimental results demonstrated the employed gesture technology has significant potential for enabling bidirectional Soldier-robot team dialogue based on the high classification accuracy and minimal training required to perform gesture commands.

Verification of a Proposed Clinical Electroacoustic Test Protocol for Personal Digital Modulation Receivers Coupled to Cochlear Implant Sound Processors.

PubMed

Nair, Erika L; Sousa, Rhonda; Wannagot, Shannon

Guidelines established by the AAA currently recommend behavioral testing when fitting frequency modulated (FM) systems to individuals with cochlear implants (CIs). A protocol for completing electroacoustic measures has not yet been validated for personal FM systems or digital modulation (DM) systems coupled to CI sound processors. In response, some professionals have used or altered the AAA electroacoustic verification steps for fitting FM systems to hearing aids when fitting FM systems to CI sound processors. More recently steps were outlined in a proposed protocol. The purpose of this research is to review and compare the electroacoustic test measures outlined in a 2013 article by Schafer and colleagues in the Journal of the American Academy of Audiology titled "A Proposed Electroacoustic Test Protocol for Personal FM Receivers Coupled to Cochlear Implant Sound Processors" to the AAA electroacoustic verification steps for fitting FM systems to hearing aids when fitting DM systems to CI users. Electroacoustic measures were conducted on 71 CI sound processors and Phonak Roger DM systems using a proposed protocol and an adapted AAA protocol. Phonak's recommended default receiver gain setting was used for each CI sound processor manufacturer and adjusted if necessary to achieve transparency. Electroacoustic measures were conducted on Cochlear and Advanced Bionics (AB) sound processors. In this study, 28 Cochlear Nucleus 5/CP810 sound processors, 26 Cochlear Nucleus 6/CP910 sound processors, and 17 AB Naida CI Q70 sound processors were coupled in various combinations to Phonak Roger DM dedicated receivers (25 Phonak Roger 14 receivers-Cochlear dedicated receiver-and 9 Phonak Roger 17 receivers-AB dedicated receiver) and 20 Phonak Roger Inspiro transmitters. Employing both the AAA and the Schafer et al protocols, electroacoustic measurements were conducted with the Audioscan Verifit in a clinical setting on 71 CI sound processors and Phonak Roger DM systems to determine transparency and verify FM advantage, comparing speech inputs (65 dB SPL) in an effort to achieve equal outputs. If transparency was not achieved at Phonak's recommended default receiver gain, adjustments were made to the receiver gain. The integrity of the signal was monitored with the appropriate manufacturer's monitor earphones. Using the AAA hearing aid protocol, 50 of the 71 CI sound processors achieved transparency, and 59 of the 71 CI sound processors achieved transparency when using the proposed protocol at Phonak's recommended default receiver gain. After the receiver gain was adjusted, 3 of 21 CI sound processors still did not meet transparency using the AAA protocol, and 2 of 12 CI sound processors still did not meet transparency using the Schafer et al proposed protocol. Both protocols were shown to be effective in taking reliable electroacoustic measurements and demonstrate transparency. Both protocols are felt to be clinically feasible and to address the needs of populations that are unable to reliably report regarding the integrity of their personal DM systems. American Academy of Audiology

Military and Government Applications of Human-Machine Communication by Voice

NASA Astrophysics Data System (ADS)

Weinstein, Clifford J.

1995-10-01

This paper describes a range of opportunities for military and government applications of human-machine communication by voice, based on visits and contacts with numerous user organizations in the United States. The applications include some that appear to be feasible by careful integration of current state-of-the-art technology and others that will require a varying mix of advances in speech technology and in integration of the technology into applications environments. Applications that are described include (1) speech recognition and synthesis for mobile command and control; (2) speech processing for a portable multifunction soldier's computer; (3) speech- and language-based technology for naval combat team tactical training; (4) speech technology for command and control on a carrier flight deck; (5) control of auxiliary systems, and alert and warning generation, in fighter aircraft and helicopters; and (6) voice check-in, report entry, and communication for law enforcement agents or special forces. A phased approach for transfer of the technology into applications is advocated, where integration of applications systems is pursued in parallel with advanced research to meet future needs.

Extraction of Volatiles from Regolith or Soil on Mars, the Moon, and Asteroids

NASA Technical Reports Server (NTRS)

Linne, Diane; Kleinhenz, Julie; Trunek, Andrew; Hoffman, Stephen; Collins, Jacob

2017-01-01

NASA's Advanced Exploration Systems ISRU Technology Project is evaluating concepts to extract water from all resource types Near-term objectives: Produce high-fidelity mass, power, and volume estimates for mining and processing systems Identify critical challenges for development focus Begin demonstration of component and subsystem technologies in relevant environment Several processor types: Closed processors either partially or completely sealed during processing Open air processors operates at Mars ambient conditions In-situ processors Extract product directly without excavation of raw resource Design features Elimination of sweep gas reduces dust particles in water condensate Pressure maintained by height of soil in hopper Model developed to evaluate key design parameters Geometry: conveyor diameter, screw diameter, shaft diameter, flight spacing and pitch Operational: screw speed vs. screw length (residence time) Thermal: Heat flux, heat transfer to soil Testing to demonstrate feasibility and performance Agglomeration, clogging Pressure rise forced flow to condenser.

An efficient ASIC implementation of 16-channel on-line recursive ICA processor for real-time EEG system.

PubMed

Fang, Wai-Chi; Huang, Kuan-Ju; Chou, Chia-Ching; Chang, Jui-Chung; Cauwenberghs, Gert; Jung, Tzyy-Ping

2014-01-01

This is a proposal for an efficient very-large-scale integration (VLSI) design, 16-channel on-line recursive independent component analysis (ORICA) processor ASIC for real-time EEG system, implemented with TSMC 40 nm CMOS technology. ORICA is appropriate to be used in real-time EEG system to separate artifacts because of its highly efficient and real-time process features. The proposed ORICA processor is composed of an ORICA processing unit and a singular value decomposition (SVD) processing unit. Compared with previous work [1], this proposed ORICA processor has enhanced effectiveness and reduced hardware complexity by utilizing a deeper pipeline architecture, shared arithmetic processing unit, and shared registers. The 16-channel random signals which contain 8-channel super-Gaussian and 8-channel sub-Gaussian components are used to analyze the dependence of the source components, and the average correlation coefficient is 0.95452 between the original source signals and extracted ORICA signals. Finally, the proposed ORICA processor ASIC is implemented with TSMC 40 nm CMOS technology, and it consumes 15.72 mW at 100 MHz operating frequency.

Extended performance electric propulsion power processor design study. Volume 2: Technical summary

NASA Technical Reports Server (NTRS)

Biess, J. J.; Inouye, L. Y.; Schoenfeld, A. D.

1977-01-01

Electric propulsion power processor technology has processed during the past decade to the point that it is considered ready for application. Several power processor design concepts were evaluated and compared. Emphasis was placed on a 30 cm ion thruster power processor with a beam power rating supply of 2.2KW to 10KW for the main propulsion power stage. Extension in power processor performance were defined and were designed in sufficient detail to determine efficiency, component weight, part count, reliability and thermal control. A detail design was performed on a microprocessor as the thyristor power processor controller. A reliability analysis was performed to evaluate the effect of the control electronics redesign. Preliminary electrical design, mechanical design and thermal analysis were performed on a 6KW power transformer for the beam supply. Bi-Mod mechanical, structural and thermal control configurations were evaluated for the power processor and preliminary estimates of mechanical weight were determined.

A perspective on early commercial applications of voice-processing technology for telecommunications and aids for the handicapped.

PubMed Central

Seelbach, C

1995-01-01

The Colloquium on Human-Machine Communication by Voice highlighted the global technical community's focus on the problems and promise of voice-processing technology, particularly, speech recognition and speech synthesis. Clearly, there are many areas in both the research and development of these technologies that can be advanced significantly. However, it is also true that there are many applications of these technologies that are capable of commercialization now. Early successful commercialization of new technology is vital to ensure continuing interest in its development. This paper addresses efforts to commercialize speech technologies in two markets: telecommunications and aids for the handicapped. PMID:7479814

An Evaluation of Output Signal to Noise Ratio as a Predictor of Cochlear Implant Speech Intelligibility.

PubMed

Watkins, Greg D; Swanson, Brett A; Suaning, Gregg J

2018-02-22

Cochlear implant (CI) sound processing strategies are usually evaluated in clinical studies involving experienced implant recipients. Metrics which estimate the capacity to perceive speech for a given set of audio and processing conditions provide an alternative means to assess the effectiveness of processing strategies. The aim of this research was to assess the ability of the output signal to noise ratio (OSNR) to accurately predict speech perception. It was hypothesized that compared with the other metrics evaluated in this study (1) OSNR would have equivalent or better accuracy and (2) OSNR would be the most accurate in the presence of variable levels of speech presentation. For the first time, the accuracy of OSNR as a metric which predicts speech intelligibility was compared, in a retrospective study, with that of the input signal to noise ratio (ISNR) and the short-term objective intelligibility (STOI) metric. Because STOI measured audio quality at the input to a CI sound processor, a vocoder was applied to the sound processor output and STOI was also calculated for the reconstructed audio signal (vocoder short-term objective intelligibility [VSTOI] metric). The figures of merit calculated for each metric were Pearson correlation of the metric and a psychometric function fitted to sentence scores at each predictor value (Pearson sigmoidal correlation [PSIG]), epsilon insensitive root mean square error (RMSE*) of the psychometric function and the sentence scores, and the statistical deviance of the fitted curve to the sentence scores (D). Sentence scores were taken from three existing data sets of Australian Sentence Tests in Noise results. The AuSTIN tests were conducted with experienced users of the Nucleus CI system. The score for each sentence was the proportion of morphemes the participant correctly repeated. In data set 1, all sentences were presented at 65 dB sound pressure level (SPL) in the presence of four-talker Babble noise. Each block of sentences used an adaptive procedure, with the speech presented at a fixed level and the ISNR varied. In data set 2, sentences were presented at 65 dB SPL in the presence of stationary speech weighted noise, street-side city noise, and cocktail party noise. An adaptive ISNR procedure was used. In data set 3, sentences were presented at levels ranging from 55 to 89 dB SPL with two automatic gain control configurations and two fixed ISNRs. For data set 1, the ISNR and OSNR were equally most accurate. STOI was significantly different for deviance (p = 0.045) and RMSE* (p < 0.001). VSTOI was significantly different for RMSE* (p < 0.001). For data set 2, ISNR and OSNR had an equivalent accuracy which was significantly better than that of STOI for PSIG (p = 0.029) and VSTOI for deviance (p = 0.001), RMSE*, and PSIG (both p < 0.001). For data set 3, OSNR was the most accurate metric and was significantly more accurate than VSTOI for deviance, RMSE*, and PSIG (all p < 0.001). ISNR and STOI were unable to predict the sentence scores for this data set. The study results supported the hypotheses. OSNR was found to have an accuracy equivalent to or better than ISNR, STOI, and VSTOI for tests conducted at a fixed presentation level and variable ISNR. OSNR was a more accurate metric than VSTOI for tests with fixed ISNRs and variable presentation levels. Overall, OSNR was the most accurate metric across the three data sets. OSNR holds promise as a prediction metric which could potentially improve the effectiveness of sound processor research and CI fitting.

Integrating a Natural Language Message Pre-Processor with UIMA

DTIC Science & Technology

2008-01-01

Carnegie Mellon Language Technologies Institute NL Message Preprocessing with UIMA Copyright © 2008, Carnegie Mellon. All Rights Reserved...Integrating a Natural Language Message Pre-Processor with UIMA Eric Nyberg, Eric Riebling, Richard C. Wang & Robert Frederking Language Technologies Institute...with UIMA 5a. CONTRACT NUMBER 5b. GRANT NUMBER 5c. PROGRAM ELEMENT NUMBER 6. AUTHOR(S) 5d. PROJECT NUMBER 5e. TASK NUMBER 5f. WORK UNIT NUMBER

Developing and Evaluating an Oral Skills Training Website Supported by Automatic Speech Recognition Technology

ERIC Educational Resources Information Center

Chen, Howard Hao-Jan

2011-01-01

Oral communication ability has become increasingly important to many EFL students. Several commercial software programs based on automatic speech recognition (ASR) technologies are available but their prices are not affordable for many students. This paper will demonstrate how the Microsoft Speech Application Software Development Kit (SASDK), a…

Computer-assisted CI fitting: Is the learning capacity of the intelligent agent FOX beneficial for speech understanding?

PubMed

Meeuws, Matthias; Pascoal, David; Bermejo, Iñigo; Artaso, Miguel; De Ceulaer, Geert; Govaerts, Paul J

2017-07-01

The software application FOX ('Fitting to Outcome eXpert') is an intelligent agent to assist in the programing of cochlear implant (CI) processors. The current version utilizes a mixture of deterministic and probabilistic logic which is able to improve over time through a learning effect. This study aimed at assessing whether this learning capacity yields measurable improvements in speech understanding. A retrospective study was performed on 25 consecutive CI recipients with a median CI use experience of 10 years who came for their annual CI follow-up fitting session. All subjects were assessed by means of speech audiometry with open set monosyllables at 40, 55, 70, and 85 dB SPL in quiet with their home MAP. Other psychoacoustic tests were executed depending on the audiologist's clinical judgment. The home MAP and the corresponding test results were entered into FOX. If FOX suggested to make MAP changes, they were implemented and another speech audiometry was performed with the new MAP. FOX suggested MAP changes in 21 subjects (84%). The within-subject comparison showed a significant median improvement of 10, 3, 1, and 7% at 40, 55, 70, and 85 dB SPL, respectively. All but two subjects showed an instantaneous improvement in their mean speech audiometric score. Persons with long-term CI use, who received a FOX-assisted CI fitting at least 6 months ago, display improved speech understanding after MAP modifications, as recommended by the current version of FOX. This can be explained only by intrinsic improvements in FOX's algorithms, as they have resulted from learning. This learning is an inherent feature of artificial intelligence and it may yield measurable benefit in speech understanding even in long-term CI recipients.

Spectro-temporal cues enhance modulation sensitivity in cochlear implant users.

PubMed

Zheng, Yi; Escabí, Monty; Litovsky, Ruth Y

2017-08-01

Although speech understanding is highly variable amongst cochlear implants (CIs) subjects, the remarkably high speech recognition performance of many CI users is unexpected and not well understood. Numerous factors, including neural health and degradation of the spectral information in the speech signal of CIs, likely contribute to speech understanding. We studied the ability to use spectro-temporal modulations, which may be critical for speech understanding and discrimination, and hypothesize that CI users adopt a different perceptual strategy than normal-hearing (NH) individuals, whereby they rely more heavily on joint spectro-temporal cues to enhance detection of auditory cues. Modulation detection sensitivity was studied in CI users and NH subjects using broadband "ripple" stimuli that were modulated spectrally, temporally, or jointly, i.e., spectro-temporally. The spectro-temporal modulation transfer functions of CI users and NH subjects was decomposed into spectral and temporal dimensions and compared to those subjects' spectral-only and temporal-only modulation transfer functions. In CI users, the joint spectro-temporal sensitivity was better than that predicted by spectral-only and temporal-only sensitivity, indicating a heightened spectro-temporal sensitivity. Such an enhancement through the combined integration of spectral and temporal cues was not observed in NH subjects. The unique use of spectro-temporal cues by CI patients can yield benefits for use of cues that are important for speech understanding. This finding has implications for developing sound processing strategies that may rely on joint spectro-temporal modulations to improve speech comprehension of CI users, and the findings of this study may be valuable for developing clinical assessment tools to optimize CI processor performance. Copyright © 2017 Elsevier B.V. All rights reserved.

JPRS Report, Science & Technology, China, High-Performance Computer Systems

DTIC Science & Technology

1992-10-28

microprocessor array The microprocessor array in the AP85 system is com- posed of 16 completely identical array element micro - processors . Each array element...microprocessors and capable of host machine reading and writing. The memory capacity of the array element micro - processors as a whole can be expanded...transmission functions to carry out data transmission from array element micro - processor to array element microprocessor, from array element

Demonstration of two-qubit algorithms with a superconducting quantum processor.

PubMed

DiCarlo, L; Chow, J M; Gambetta, J M; Bishop, Lev S; Johnson, B R; Schuster, D I; Majer, J; Blais, A; Frunzio, L; Girvin, S M; Schoelkopf, R J

2009-07-09

Quantum computers, which harness the superposition and entanglement of physical states, could outperform their classical counterparts in solving problems with technological impact-such as factoring large numbers and searching databases. A quantum processor executes algorithms by applying a programmable sequence of gates to an initialized register of qubits, which coherently evolves into a final state containing the result of the computation. Building a quantum processor is challenging because of the need to meet simultaneously requirements that are in conflict: state preparation, long coherence times, universal gate operations and qubit readout. Processors based on a few qubits have been demonstrated using nuclear magnetic resonance, cold ion trap and optical systems, but a solid-state realization has remained an outstanding challenge. Here we demonstrate a two-qubit superconducting processor and the implementation of the Grover search and Deutsch-Jozsa quantum algorithms. We use a two-qubit interaction, tunable in strength by two orders of magnitude on nanosecond timescales, which is mediated by a cavity bus in a circuit quantum electrodynamics architecture. This interaction allows the generation of highly entangled states with concurrence up to 94 per cent. Although this processor constitutes an important step in quantum computing with integrated circuits, continuing efforts to increase qubit coherence times, gate performance and register size will be required to fulfil the promise of a scalable technology.

Right-Brain/Left-Brain Integrated Associative Processor Employing Convertible Multiple-Instruction-Stream Multiple-Data-Stream Elements

NASA Astrophysics Data System (ADS)

Hayakawa, Hitoshi; Ogawa, Makoto; Shibata, Tadashi

2005-04-01

A very large scale integrated circuit (VLSI) architecture for a multiple-instruction-stream multiple-data-stream (MIMD) associative processor has been proposed. The processor employs an architecture that enables seamless switching from associative operations to arithmetic operations. The MIMD element is convertible to a regular central processing unit (CPU) while maintaining its high performance as an associative processor. Therefore, the MIMD associative processor can perform not only on-chip perception, i.e., searching for the vector most similar to an input vector throughout the on-chip cache memory, but also arithmetic and logic operations similar to those in ordinary CPUs, both simultaneously in parallel processing. Three key technologies have been developed to generate the MIMD element: associative-operation-and-arithmetic-operation switchable calculation units, a versatile register control scheme within the MIMD element for flexible operations, and a short instruction set for minimizing the memory size for program storage. Key circuit blocks were designed and fabricated using 0.18 μm complementary metal-oxide-semiconductor (CMOS) technology. As a result, the full-featured MIMD element is estimated to be 3 mm2, showing the feasibility of an 8-parallel-MIMD-element associative processor in a single chip of 5 mm× 5 mm.

Automatic speech recognition (ASR) based approach for speech therapy of aphasic patients: A review

NASA Astrophysics Data System (ADS)

Jamal, Norezmi; Shanta, Shahnoor; Mahmud, Farhanahani; Sha'abani, MNAH

2017-09-01

This paper reviews the state-of-the-art an automatic speech recognition (ASR) based approach for speech therapy of aphasic patients. Aphasia is a condition in which the affected person suffers from speech and language disorder resulting from a stroke or brain injury. Since there is a growing body of evidence indicating the possibility of improving the symptoms at an early stage, ASR based solutions are increasingly being researched for speech and language therapy. ASR is a technology that transfers human speech into transcript text by matching with the system's library. This is particularly useful in speech rehabilitation therapies as they provide accurate, real-time evaluation for speech input from an individual with speech disorder. ASR based approaches for speech therapy recognize the speech input from the aphasic patient and provide real-time feedback response to their mistakes. However, the accuracy of ASR is dependent on many factors such as, phoneme recognition, speech continuity, speaker and environmental differences as well as our depth of knowledge on human language understanding. Hence, the review examines recent development of ASR technologies and its performance for individuals with speech and language disorders.

A Class of CFAR Detectors Implemented in the SAR-GMTI Processor gmtipro2: Mathematical Formulation of the Algorithms

DTIC Science & Technology

2015-02-01

Right of Canada as represented by the Minister of National Defence, 2015 c© Sa Majesté la Reine (en droit du Canada), telle que représentée par le...References [1] Chiu, S. (2010), Moving target parameter estimation for RADARSAT-2 Moving Object Detection EXperiment (MODEX), International Journal of...of multiple sinusoids in noise, In Proceedings. (ICASSP ’01). 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing, Vol. 5

Binaural hearing with electrical stimulation

PubMed Central

Kan, Alan; Litovsky, Ruth Y.

2014-01-01

Bilateral cochlear implantation is becoming a standard of care in many clinics. While much benefit has been shown through bilateral implantation, patients who have bilateral cochlear implants (CIs) still do not perform as well as normal hearing listeners in sound localization and understanding speech in noisy environments. This difference in performance can arise from a number of different factors, including the areas of hardware and engineering, surgical precision and pathology of the auditory system in deaf persons. While surgical precision and individual pathology are factors that are beyond careful control, improvements can be made in the areas of clinical practice and the engineering of binaural speech processors. These improvements should be grounded in a good understanding of the sensitivities of bilateral CI patients to the acoustic binaural cues that are important to normal hearing listeners for sound localization and speech in noise understanding. To this end, we review the current state-of-the-art in the understanding of the sensitivities of bilateral CI patients to binaural cues in electric hearing, and highlight the important issues and challenges as they relate to clinical practice and the development of new binaural processing strategies. PMID:25193553

Compiling for Application Specific Computational Acceleration in Reconfigurable Architectures Final Report CRADA No. TSB-2033-01

DOE Office of Scientific and Technical Information (OSTI.GOV)

De Supinski, B.; Caliga, D.

2017-09-28

The primary objective of this project was to develop memory optimization technology to efficiently deliver data to, and distribute data within, the SRC-6's Field Programmable Gate Array- ("FPGA") based Multi-Adaptive Processors (MAPs). The hardware/software approach was to explore efficient MAP configurations and generate the compiler technology to exploit those configurations. This memory accessing technology represents an important step towards making reconfigurable symmetric multi-processor (SMP) architectures that will be a costeffective solution for large-scale scientific computing.

Binaural release from masking with single- and multi-electrode stimulation in children with cochlear implantsa)

PubMed Central

Todd, Ann E.; Goupell, Matthew J.; Litovsky, Ruth Y.

2016-01-01

Cochlear implants (CIs) provide children with access to speech information from a young age. Despite bilateral cochlear implantation becoming common, use of spatial cues in free field is smaller than in normal-hearing children. Clinically fit CIs are not synchronized across the ears; thus binaural experiments must utilize research processors that can control binaural cues with precision. Research to date has used single pairs of electrodes, which is insufficient for representing speech. Little is known about how children with bilateral CIs process binaural information with multi-electrode stimulation. Toward the goal of improving binaural unmasking of speech, this study evaluated binaural unmasking with multi- and single-electrode stimulation. Results showed that performance with multi-electrode stimulation was similar to the best performance with single-electrode stimulation. This was similar to the pattern of performance shown by normal-hearing adults when presented an acoustic CI simulation. Diotic and dichotic signal detection thresholds of the children with CIs were similar to those of normal-hearing children listening to a CI simulation. The magnitude of binaural unmasking was not related to whether the children with CIs had good interaural time difference sensitivity. Results support the potential for benefits from binaural hearing and speech unmasking in children with bilateral CIs. PMID:27475132

Binaural release from masking with single- and multi-electrode stimulation in children with cochlear implants.

PubMed

Todd, Ann E; Goupell, Matthew J; Litovsky, Ruth Y

2016-07-01

Cochlear implants (CIs) provide children with access to speech information from a young age. Despite bilateral cochlear implantation becoming common, use of spatial cues in free field is smaller than in normal-hearing children. Clinically fit CIs are not synchronized across the ears; thus binaural experiments must utilize research processors that can control binaural cues with precision. Research to date has used single pairs of electrodes, which is insufficient for representing speech. Little is known about how children with bilateral CIs process binaural information with multi-electrode stimulation. Toward the goal of improving binaural unmasking of speech, this study evaluated binaural unmasking with multi- and single-electrode stimulation. Results showed that performance with multi-electrode stimulation was similar to the best performance with single-electrode stimulation. This was similar to the pattern of performance shown by normal-hearing adults when presented an acoustic CI simulation. Diotic and dichotic signal detection thresholds of the children with CIs were similar to those of normal-hearing children listening to a CI simulation. The magnitude of binaural unmasking was not related to whether the children with CIs had good interaural time difference sensitivity. Results support the potential for benefits from binaural hearing and speech unmasking in children with bilateral CIs.

Comprehension of synthetic speech and digitized natural speech by adults with aphasia.

PubMed

Hux, Karen; Knollman-Porter, Kelly; Brown, Jessica; Wallace, Sarah E

2017-09-01

Using text-to-speech technology to provide simultaneous written and auditory content presentation may help compensate for chronic reading challenges if people with aphasia can understand synthetic speech output; however, inherent auditory comprehension challenges experienced by people with aphasia may make understanding synthetic speech difficult. This study's purpose was to compare the preferences and auditory comprehension accuracy of people with aphasia when listening to sentences generated with digitized natural speech, Alex synthetic speech (i.e., Macintosh platform), or David synthetic speech (i.e., Windows platform). The methodology required each of 20 participants with aphasia to select one of four images corresponding in meaning to each of 60 sentences comprising three stimulus sets. Results revealed significantly better accuracy given digitized natural speech than either synthetic speech option; however, individual participant performance analyses revealed three patterns: (a) comparable accuracy regardless of speech condition for 30% of participants, (b) comparable accuracy between digitized natural speech and one, but not both, synthetic speech option for 45% of participants, and (c) greater accuracy with digitized natural speech than with either synthetic speech option for remaining participants. Ranking and Likert-scale rating data revealed a preference for digitized natural speech and David synthetic speech over Alex synthetic speech. Results suggest many individuals with aphasia can comprehend synthetic speech options available on popular operating systems. Further examination of synthetic speech use to support reading comprehension through text-to-speech technology is thus warranted. Copyright © 2017 Elsevier Inc. All rights reserved.

Military and government applications of human-machine communication by voice.

PubMed Central

Weinstein, C J

1995-01-01

This paper describes a range of opportunities for military and government applications of human-machine communication by voice, based on visits and contacts with numerous user organizations in the United States. The applications include some that appear to be feasible by careful integration of current state-of-the-art technology and others that will require a varying mix of advances in speech technology and in integration of the technology into applications environments. Applications that are described include (1) speech recognition and synthesis for mobile command and control; (2) speech processing for a portable multifunction soldier's computer; (3) speech- and language-based technology for naval combat team tactical training; (4) speech technology for command and control on a carrier flight deck; (5) control of auxiliary systems, and alert and warning generation, in fighter aircraft and helicopters; and (6) voice check-in, report entry, and communication for law enforcement agents or special forces. A phased approach for transfer of the technology into applications is advocated, where integration of applications systems is pursued in parallel with advanced research to meet future needs. Images Fig. 1 Fig. 2 Fig. 3 Fig. 4 Fig. 5 Fig. 6 PMID:7479718

Early Student Support for Application of Advanced Multi-Core Processor Technologies to Oceanographic Research

DTIC Science & Technology

2016-05-07

REPORT DOCUMENTATION PAGE I . ... ... .. . ,...,.., ............. OMB No. 0704-0188 The public reporting burden for this collection of...Student Support for Appl ication of Advanced Multi- Core Processor N00014-12-1-0298 Technologies to Oceanographic Research Sb. GRANT NUMBER Sc...communications protocols (i.e. UART, I2C, and SPI), through the , ’ . handing off of the data to the server APis. By providing a common set of tools

SAM: speech-aware applications in medicine to support structured data entry.

PubMed Central

Wormek, A. K.; Ingenerf, J.; Orthner, H. F.

1997-01-01

In the last two years, improvement in speech recognition technology has directed the medical community's interest to porting and using such innovations in clinical systems. The acceptance of speech recognition systems in clinical domains increases with recognition speed, large medical vocabulary, high accuracy, continuous speech recognition, and speaker independence. Although some commercial speech engines approach these requirements, the greatest benefit can be achieved in adapting a speech recognizer to a specific medical application. The goals of our work are first, to develop a speech-aware core component which is able to establish connections to speech recognition engines of different vendors. This is realized in SAM. Second, with applications based on SAM we want to support the physician in his/her routine clinical care activities. Within the STAMP project (STAndardized Multimedia report generator in Pathology), we extend SAM by combining a structured data entry approach with speech recognition technology. Another speech-aware application in the field of Diabetes care is connected to a terminology server. The server delivers a controlled vocabulary which can be used for speech recognition. PMID:9357730

Nursing acceptance of a speech-input interface: a preliminary investigation.

PubMed

Dillon, T W; McDowell, D; Norcio, A F; DeHaemer, M J

1994-01-01

Many new technologies are being developed to improve the efficiency and productivity of nursing staffs. User acceptance is a key to the success of these technologies. In this article, the authors present a discussion of nursing acceptance of computer systems, review the basic design issues for creating a speech-input interface, and report preliminary findings of a study of nursing acceptance of a prototype speech-input interface. Results of the study showed that the 19 nursing subjects expressed acceptance of the prototype speech-input interface.

Effects of programming threshold and maplaw settings on acoustic thresholds and speech discrimination with the MED-EL COMBI 40+ cochlear implant.

PubMed

Boyd, Paul J

2006-12-01

The principal task in the programming of a cochlear implant (CI) speech processor is the setting of the electrical dynamic range (output) for each electrode, to ensure that a comfortable loudness percept is obtained for a range of input levels. This typically involves separate psychophysical measurement of electrical threshold ([theta] e) and upper tolerance levels using short current bursts generated by the fitting software. Anecdotal clinical experience and some experimental studies suggest that the measurement of [theta]e is relatively unimportant and that the setting of upper tolerance limits is more critical for processor programming. The present study aims to test this hypothesis and examines in detail how acoustic thresholds and speech recognition are affected by setting of the lower limit of the output ("Programming threshold" or "PT") to understand better the influence of this parameter and how it interacts with certain other programming parameters. Test programs (maps) were generated with PT set to artificially high and low values and tested on users of the MED-EL COMBI 40+ CI system. Acoustic thresholds and speech recognition scores (sentence tests) were measured for each of the test maps. Acoustic thresholds were also measured using maps with a range of output compression functions ("maplaws"). In addition, subjective reports were recorded regarding the presence of "background threshold stimulation" which is occasionally reported by CI users if PT is set to relatively high values when using the CIS strategy. Manipulation of PT was found to have very little effect. Setting PT to minimum produced a mean 5 dB (S.D. = 6.25) increase in acoustic thresholds, relative to thresholds with PT set normally, and had no statistically significant effect on speech recognition scores on a sentence test. On the other hand, maplaw setting was found to have a significant effect on acoustic thresholds (raised as maplaw is made more linear), which provides some theoretical explanation as to why PT has little effect when using the default maplaw of c = 500. Subjective reports of background threshold stimulation showed that most users could perceive a relatively loud auditory percept, in the absence of microphone input, when PT was set to double the behaviorally measured electrical thresholds ([theta]e), but that this produced little intrusion when microphone input was present. The results of these investigations have direct clinical relevance, showing that setting of PT is indeed relatively unimportant in terms of speech discrimination, but that it is worth ensuring that PT is not set excessively high, as this can produce distracting background stimulation. Indeed, it may even be set to minimum values without deleterious effect.

Effect of different signal-processing options on speech-in-noise recognition for cochlear implant recipients with the cochlear CP810 speech processor.

PubMed

Potts, Lisa G; Kolb, Kelly A

2014-04-01

Difficulty understanding speech in the presence of background noise is a common report among cochlear implant (CI) recipients. Several speech-processing options designed to improve speech recognition, especially in noise, are currently available in the Cochlear Nucleus CP810 speech processor. These include adaptive dynamic range optimization (ADRO), autosensitivity control (ASC), Beam, and Zoom. The purpose of this study was to evaluate CI recipients' speech-in-noise recognition to determine which currently available processing option or options resulted in best performance in a simulated restaurant environment. Experimental study with one study group. The independent variable was speech-processing option, and the dependent variable was the reception threshold for sentences score. Thirty-two adult CI recipients. Eight processing options were tested: Beam, Beam + ASC, Beam + ADRO, Beam + ASC + ADRO, Zoom, Zoom + ASC, Zoom + ADRO, and Zoom + ASC + ADRO. Participants repeated Hearing in Noise Test sentences presented at a 0° azimuth, with R-Space restaurant noise presented from a 360° eight-loudspeaker array at 70 dB sound pressure level. A one-way repeated-measures analysis of variance was used to analyze differences in Beam options, Zoom options, and Beam versus Zoom options. Among the Beam options, Beam + ADRO was significantly poorer than Beam only, Beam + ASC, and Beam + ASC + ADRO. A 1.6-dB difference was observed between the best (Beam only) and poorest (Beam + ADRO) options. Among the Zoom options, Zoom only and Zoom + ADRO were significantly poorer than Zoom + ASC. A 2.2-dB difference was observed between the best (Zoom + ASC) and poorest (Zoom only) options. The comparison between Beam and Zoom options showed one significant difference, with Zoom only significantly poorer than Beam only. No significant difference was found between the other Beam and Zoom options (Beam + ASC vs Zoom + ASC, Beam + ADRO vs Zoom + ADRO, and Beam + ASC + ADRO vs Zoom + ASC + ADRO). The best processing option varied across subjects, with an almost equal number of participants performing best with a Beam option (n = 15) compared with a Zoom option (n = 17). There were no significant demographic or audiological moderating variables for any option. The results showed no significant differences between adaptive directionality (Beam) and fixed directionality (Zoom) when ASC was active in the R-Space environment. This finding suggests that noise-reduction processing is extremely valuable in loud semidiffuse environments in which the effectiveness of directional filtering might be diminished. However, there was no significant difference between the Beam-only and Beam + ASC options, which is most likely related to the additional noise cancellation performed by the Beam option (i.e., two-stage directional filtering and noise cancellation). In addition, the processing options with ADRO resulted in the poorest performances. This could be related to how the CI recipients were programmed or the loud noise level used in this study. The best processing option varied across subjects, but the majority performed best with directional filtering (Beam or Zoom) in combination with ASC. Therefore in a loud semidiffuse environment, the use of either Beam + ASC or Zoom + ASC is recommended. American Academy of Audiology.

Direct RF A-O Processor Spectrum Analyzer.

DTIC Science & Technology

1981-08-01

The primary objective was to develop and demonstrate design approach, along with the associated processing technologies, for a wideband acousto optic Bragg...cell spectrum analyzer. The signal processor used to demonstrate feasibility of the technical approach consisted of two bulk wave acousto optic deflectors

A light hydrocarbon fuel processor producing high-purity hydrogen

NASA Astrophysics Data System (ADS)

Löffler, Daniel G.; Taylor, Kyle; Mason, Dylan

This paper discusses the design process and presents performance data for a dual fuel (natural gas and LPG) fuel processor for PEM fuel cells delivering between 2 and 8 kW electric power in stationary applications. The fuel processor resulted from a series of design compromises made to address different design constraints. First, the product quality was selected; then, the unit operations needed to achieve that product quality were chosen from the pool of available technologies. Next, the specific equipment needed for each unit operation was selected. Finally, the unit operations were thermally integrated to achieve high thermal efficiency. Early in the design process, it was decided that the fuel processor would deliver high-purity hydrogen. Hydrogen can be separated from other gases by pressure-driven processes based on either selective adsorption or permeation. The pressure requirement made steam reforming (SR) the preferred reforming technology because it does not require compression of combustion air; therefore, steam reforming is more efficient in a high-pressure fuel processor than alternative technologies like autothermal reforming (ATR) or partial oxidation (POX), where the combustion occurs at the pressure of the process stream. A low-temperature pre-reformer reactor is needed upstream of a steam reformer to suppress coke formation; yet, low temperatures facilitate the formation of metal sulfides that deactivate the catalyst. For this reason, a desulfurization unit is needed upstream of the pre-reformer. Hydrogen separation was implemented using a palladium alloy membrane. Packed beds were chosen for the pre-reformer and reformer reactors primarily because of their low cost, relatively simple operation and low maintenance. Commercial, off-the-shelf balance of plant (BOP) components (pumps, valves, and heat exchangers) were used to integrate the unit operations. The fuel processor delivers up to 100 slm hydrogen >99.9% pure with <1 ppm CO, <3 ppm CO 2. The thermal efficiency is better than 67% operating at full load. This fuel processor has been integrated with a 5-kW fuel cell producing electricity and hot water.

Contributions of speech science to the technology of man-machine voice interactions

NASA Technical Reports Server (NTRS)

Lea, Wayne A.

1977-01-01

Research in speech understanding was reviewed. Plans which include prosodics research, phonological rules for speech understanding systems, and continued interdisciplinary phonetics research are discussed. Improved acoustic phonetic analysis capabilities in speech recognizers are suggested.

The precision-processing subsystem for the Earth Resources Technology Satellite.

NASA Technical Reports Server (NTRS)

Chapelle, W. E.; Bybee, J. E.; Bedross, G. M.

1972-01-01

Description of the precision processor, a subsystem in the image-processing system for the Earth Resources Technology Satellite (ERTS). This processor is a special-purpose image-measurement and printing system, designed to process user-selected bulk images to produce 1:1,000,000-scale film outputs and digital image data, presented in a Universal-Transverse-Mercator (UTM) projection. The system will remove geometric and radiometric errors introduced by the ERTS multispectral sensors and by the bulk-processor electron-beam recorder. The geometric transformations required for each input scene are determined by resection computations based on reseau measurements and image comparisons with a special ground-control base contained within the system; the images are then printed and digitized by electronic image-transfer techniques.

Speech Perception for Adult Cochlear Implant Recipients in a Realistic Background Noise: Effectiveness of Preprocessing Strategies and External Options for Improving Speech Recognition in Noise

PubMed Central

Gifford, René H.; Revit, Lawrence J.

2014-01-01

Background Although cochlear implant patients are achieving increasingly higher levels of performance, speech perception in noise continues to be problematic. The newest generations of implant speech processors are equipped with preprocessing and/or external accessories that are purported to improve listening in noise. Most speech perception measures in the clinical setting, however, do not provide a close approximation to real-world listening environments. Purpose To assess speech perception for adult cochlear implant recipients in the presence of a realistic restaurant simulation generated by an eight-loudspeaker (R-SPACE™) array in order to determine whether commercially available preprocessing strategies and/or external accessories yield improved sentence recognition in noise. Research Design Single-subject, repeated-measures design with two groups of participants: Advanced Bionics and Cochlear Corporation recipients. Study Sample Thirty-four subjects, ranging in age from 18 to 90 yr (mean 54.5 yr), participated in this prospective study. Fourteen subjects were Advanced Bionics recipients, and 20 subjects were Cochlear Corporation recipients. Intervention Speech reception thresholds (SRTs) in semidiffuse restaurant noise originating from an eight-loudspeaker array were assessed with the subjects’ preferred listening programs as well as with the addition of either Beam™ preprocessing (Cochlear Corporation) or the T-Mic® accessory option (Advanced Bionics). Data Collection and Analysis In Experiment 1, adaptive SRTs with the Hearing in Noise Test sentences were obtained for all 34 subjects. For Cochlear Corporation recipients, SRTs were obtained with their preferred everyday listening program as well as with the addition of Focus preprocessing. For Advanced Bionics recipients, SRTs were obtained with the integrated behind-the-ear (BTE) mic as well as with the T-Mic. Statistical analysis using a repeated-measures analysis of variance (ANOVA) evaluated the effects of the preprocessing strategy or external accessory in reducing the SRT in noise. In addition, a standard t-test was run to evaluate effectiveness across manufacturer for improving the SRT in noise. In Experiment 2, 16 of the 20 Cochlear Corporation subjects were reassessed obtaining an SRT in noise using the manufacturer-suggested “Everyday,” “Noise,” and “Focus” preprocessing strategies. A repeated-measures ANOVA was employed to assess the effects of preprocessing. Results The primary findings were (i) both Noise and Focus preprocessing strategies (Cochlear Corporation) significantly improved the SRT in noise as compared to Everyday preprocessing, (ii) the T-Mic accessory option (Advanced Bionics) significantly improved the SRT as compared to the BTE mic, and (iii) Focus preprocessing and the T-Mic resulted in similar degrees of improvement that were not found to be significantly different from one another. Conclusion Options available in current cochlear implant sound processors are able to significantly improve speech understanding in a realistic, semidiffuse noise with both Cochlear Corporation and Advanced Bionics systems. For Cochlear Corporation recipients, Focus preprocessing yields the best speech-recognition performance in a complex listening environment; however, it is recommended that Noise preprocessing be used as the new default for everyday listening environments to avoid the need for switching programs throughout the day. For Advanced Bionics recipients, the T-Mic offers significantly improved performance in noise and is recommended for everyday use in all listening environments. PMID:20807480

Speech recognition: how good is good enough?

PubMed

Krohn, Richard

2002-03-01

Since its infancy in the early 1990s, the technology of speech recognition has undergone a rapid evolution. Not only has the reliability of the programming improved dramatically, the return on investment has become increasingly compelling. The author describes some of the latest health care applications of speech-recognition technology, and how the next advances will be made in this area.

Onboard Radar Processing Development for Rapid Response Applications

NASA Technical Reports Server (NTRS)

Lou, Yunling; Chien, Steve; Clark, Duane; Doubleday, Josh; Muellerschoen, Ron; Wang, Charles C.

2011-01-01

We are developing onboard processor (OBP) technology to streamline data acquisition on-demand and explore the potential of the L-band SAR instrument onboard the proposed DESDynI mission and UAVSAR for rapid response applications. The technology would enable the observation and use of surface change data over rapidly evolving natural hazards, both as an aid to scientific understanding and to provide timely data to agencies responsible for the management and mitigation of natural disasters. We are adapting complex science algorithms for surface water extent to detect flooding, snow/water/ice classification to assist in transportation/ shipping forecasts, and repeat-pass change detection to detect disturbances. We are near completion of the development of a custom FPGA board to meet the specific memory and processing needs of L-band SAR processor algorithms and high speed interfaces to reformat and route raw radar data to/from the FPGA processor board. We have also developed a high fidelity Matlab model of the SAR processor that is modularized and parameterized for ease to prototype various SAR processor algorithms targeted for the FPGA. We will be testing the OBP and rapid response algorithms with UAVSAR data to determine the fidelity of the products.

Enabling Future Robotic Missions with Multicore Processors

NASA Technical Reports Server (NTRS)

Powell, Wesley A.; Johnson, Michael A.; Wilmot, Jonathan; Some, Raphael; Gostelow, Kim P.; Reeves, Glenn; Doyle, Richard J.

2011-01-01

Recent commercial developments in multicore processors (e.g. Tilera, Clearspeed, HyperX) have provided an option for high performance embedded computing that rivals the performance attainable with FPGA-based reconfigurable computing architectures. Furthermore, these processors offer more straightforward and streamlined application development by allowing the use of conventional programming languages and software tools in lieu of hardware design languages such as VHDL and Verilog. With these advantages, multicore processors can significantly enhance the capabilities of future robotic space missions. This paper will discuss these benefits, along with onboard processing applications where multicore processing can offer advantages over existing or competing approaches. This paper will also discuss the key artchitecural features of current commercial multicore processors. In comparison to the current art, the features and advancements necessary for spaceflight multicore processors will be identified. These include power reduction, radiation hardening, inherent fault tolerance, and support for common spacecraft bus interfaces. Lastly, this paper will explore how multicore processors might evolve with advances in electronics technology and how avionics architectures might evolve once multicore processors are inserted into NASA robotic spacecraft.

State University of New York Institute of Technology (SUNYIT) Summer Scholar Program

DTIC Science & Technology

2009-10-01

COVERED (From - To) March 2007 – April 2009 4 . TITLE AND SUBTITLE STATE UNIVERSITY OF NEW YORK INSTITUTE OF TECHNOLOGY (SUNYIT) SUMMER SCHOLAR...Even with access to the Arctic Regional Supercomputer Center (ARSC), evolving a 9/7 wavelet with four multi-resolution levels (MRA 4 ) involves...evaluated over the multiple processing elements in the Cell processor. It was tested on Cell processors in a Sony Playstation 3 and on an IBM QS20 blade

Impact of diet on the design of waste processors in CELSS

NASA Technical Reports Server (NTRS)

Waleh, Ahmad; Kanevsky, Valery; Nguyen, Thoi K.; Upadhye, Ravi; Wydeven, Theodore

1991-01-01

The preliminary results of a design analysis for a waste processor which employs existing technologies and takes into account the constraints of human diet are presented. The impact of diet is determined by using a model and an algorithm developed for the control and management of diet in a Controlled Ecological Life Support System (CELSS). A material and energy balance model for thermal oxidation of waste is developed which is consistent with both physical/chemical methods of incineration and supercritical water oxidation. The two models yield quantitative analysis of the diet and waste streams and the specific design parameters for waste processors, respectively. The results demonstrate that existing technologies can meet the demands of waste processing, but the choice and design of the processors or processing methods will be sensitive to the constraints of diet. The numerical examples are chosen to display the nature and extent of the gap in the available experiment information about CELSS requirements.

A High Performance VLSI Computer Architecture For Computer Graphics

NASA Astrophysics Data System (ADS)

Chin, Chi-Yuan; Lin, Wen-Tai

1988-10-01

A VLSI computer architecture, consisting of multiple processors, is presented in this paper to satisfy the modern computer graphics demands, e.g. high resolution, realistic animation, real-time display etc.. All processors share a global memory which are partitioned into multiple banks. Through a crossbar network, data from one memory bank can be broadcasted to many processors. Processors are physically interconnected through a hyper-crossbar network (a crossbar-like network). By programming the network, the topology of communication links among processors can be reconfigurated to satisfy specific dataflows of different applications. Each processor consists of a controller, arithmetic operators, local memory, a local crossbar network, and I/O ports to communicate with other processors, memory banks, and a system controller. Operations in each processor are characterized into two modes, i.e. object domain and space domain, to fully utilize the data-independency characteristics of graphics processing. Special graphics features such as 3D-to-2D conversion, shadow generation, texturing, and reflection, can be easily handled. With the current high density interconnection (MI) technology, it is feasible to implement a 64-processor system to achieve 2.5 billion operations per second, a performance needed in most advanced graphics applications.

Rapid prototyping and evaluation of programmable SIMD SDR processors in LISA

NASA Astrophysics Data System (ADS)

Chen, Ting; Liu, Hengzhu; Zhang, Botao; Liu, Dongpei

2013-03-01

With the development of international wireless communication standards, there is an increase in computational requirement for baseband signal processors. Time-to-market pressure makes it impossible to completely redesign new processors for the evolving standards. Due to its high flexibility and low power, software defined radio (SDR) digital signal processors have been proposed as promising technology to replace traditional ASIC and FPGA fashions. In addition, there are large numbers of parallel data processed in computation-intensive functions, which fosters the development of single instruction multiple data (SIMD) architecture in SDR platform. So a new way must be found to prototype the SDR processors efficiently. In this paper we present a bit-and-cycle accurate model of programmable SIMD SDR processors in a machine description language LISA. LISA is a language for instruction set architecture which can gain rapid model at architectural level. In order to evaluate the availability of our proposed processor, three common baseband functions, FFT, FIR digital filter and matrix multiplication have been mapped on the SDR platform. Analytical results showed that the SDR processor achieved the maximum of 47.1% performance boost relative to the opponent processor.

Statistical assessment of speech system performance

NASA Technical Reports Server (NTRS)

Moshier, Stephen L.

1977-01-01

Methods for the normalization of performance tests results of speech recognition systems are presented. Technological accomplishments in speech recognition systems, as well as planned research activities are described.

Methods for Trustworthy Design of On-Chip Bus Interconnect for General-Purpose Processors

DTIC Science & Technology

2012-03-01

Technology Andrew Huang, was able to test the security properties of HyperTransport bus protocol on an Xbox [20]. In his research, he was able to...TRUSTWORTHY DESIGN OF ON -CHIP BUS INTERCONNECT FOR GENERAL-PURPOSE PROCESSORS by Jay F. Elson March 2012 Thesis Advisor: Ted Huffmire Second...AND DATES COVERED Master’s Thesis 4. TITLE AND SUBTITLE Methods for Trustworthy Design of On -Chip Bus Interconnect for General-Purpose Processors 5

The application of charge-coupled device processors in automatic-control systems

NASA Technical Reports Server (NTRS)

Mcvey, E. S.; Parrish, E. A., Jr.

1977-01-01

The application of charge-coupled device (CCD) processors to automatic-control systems is suggested. CCD processors are a new form of semiconductor component with the unique ability to process sampled signals on an analog basis. Specific implementations of controllers are suggested for linear time-invariant, time-varying, and nonlinear systems. Typical processing time should be only a few microseconds. This form of technology may become competitive with microprocessors and minicomputers in addition to supplementing them.

Application of OpenCV in Asus Tinker Board for face recognition

NASA Astrophysics Data System (ADS)

Chen, Wei-Yu; Wu, Frank; Hu, Chung-Chiang

2017-06-01

The rise of the Internet of Things to promote the development of technology development board, the processor speed of operation and memory capacity increases, more and more applications, can already be completed before the data on the board computing, combined with the network to sort the information after Sent to the cloud for processing, so that the front of the development board is no longer simply retrieve the data device. This study uses Asus Tinker Board to install OpenCV for real-time face recognition and capture of the face, the acquired face to the Microsoft Cognitive Service cloud database for artificial intelligence comparison, to find out what the face now represents the mood. The face of the corresponding person name, and finally, and then through the text of Speech to read the name of the name to complete the identification of the action. This study was developed using the Asus Tinker Board, which uses ARM-based CPUs with high efficiency and low power consumption, plus improvements in memory and hardware performance for the development board.

On VLSI Design of Rank-Order Filtering using DCRAM Architecture

PubMed Central

Lin, Meng-Chun; Dung, Lan-Rong

2009-01-01

This paper addresses on VLSI design of rank-order filtering (ROF) with a maskable memory for real-time speech and image processing applications. Based on a generic bit-sliced ROF algorithm, the proposed design uses a special-defined memory, called the dual-cell random-access memory (DCRAM), to realize major operations of ROF: threshold decomposition and polarization. Using the memory-oriented architecture, the proposed ROF processor can benefit from high flexibility, low cost and high speed. The DCRAM can perform the bit-sliced read, partial write, and pipelined processing. The bit-sliced read and partial write are driven by maskable registers. With recursive execution of the bit-slicing read and partial write, the DCRAM can effectively realize ROF in terms of cost and speed. The proposed design has been implemented using TSMC 0.18 μm 1P6M technology. As shown in the result of physical implementation, the core size is 356.1 × 427.7μm2 and the VLSI implementation of ROF can operate at 256 MHz for 1.8V supply. PMID:19865599

Research in speech communication.

PubMed

Flanagan, J

1995-10-24

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker.

Communication Supports for People with Motor Speech Disorders

ERIC Educational Resources Information Center

Hanson, Elizabeth K.; Fager, Susan K.

2017-01-01

Communication supports for people with motor speech disorders can include strategies and technologies to supplement natural speech efforts, resolve communication breakdowns, and replace natural speech when necessary to enhance participation in all communicative contexts. This article emphasizes communication supports that can enhance…

Automatic feedback to promote safe walking and speech loudness control in persons with multiple disabilities: two single-case studies.

PubMed

Lancioni, Giulio E; Singh, Nirbhay N; O'Reilly, Mark F; Green, Vanessa A; Alberti, Gloria; Boccasini, Adele; Smaldone, Angela; Oliva, Doretta; Bosco, Andrea

2014-08-01

Assessing automatic feedback technologies to promote safe travel and speech loudness control in two men with multiple disabilities, respectively. The men were involved in two single-case studies. In Study I, the technology involved a microprocessor, two photocells, and a verbal feedback device. The man received verbal alerting/feedback when the photocells spotted an obstacle in front of him. In Study II, the technology involved a sound-detecting unit connected to a throat and an airborne microphone, and to a vibration device. Vibration occurred when the man's speech loudness exceeded a preset level. The man included in Study I succeeded in using the automatic feedback in substitution of caregivers' alerting/feedback for safe travel. The man of Study II used the automatic feedback to successfully reduce his speech loudness. Automatic feedback can be highly effective in helping persons with multiple disabilities improve their travel and speech performance.

The feasibility of miniaturizing the versatile portable speech prosthesis: A market survey of commercial products

NASA Technical Reports Server (NTRS)

Walklet, T.

1981-01-01

The feasibility of a miniature versatile portable speech prosthesis (VPSP) was analyzed and information on its potential users and on other similar devices was collected. The VPSP is a device that incorporates speech synthesis technology. The objective is to provide sufficient information to decide whether there is valuable technology to contribute to the miniaturization of the VPSP. The needs of potential users are identified, the development status of technologies similar or related to those used in the VPSP are evaluated. The VPSP, a computer based speech synthesis system fits on a wheelchair. The purpose was to produce a device that provides communication assistance in educational, vocational, and social situations to speech impaired individuals. It is expected that the VPSP can be a valuable aid for persons who are also motor impaired, which explains the placement of the system on a wheelchair.

Processing of speech signals for physical and sensory disabilities.

PubMed Central

Levitt, H

1995-01-01

Assistive technology involving voice communication is used primarily by people who are deaf, hard of hearing, or who have speech and/or language disabilities. It is also used to a lesser extent by people with visual or motor disabilities. A very wide range of devices has been developed for people with hearing loss. These devices can be categorized not only by the modality of stimulation [i.e., auditory, visual, tactile, or direct electrical stimulation of the auditory nerve (auditory-neural)] but also in terms of the degree of speech processing that is used. At least four such categories can be distinguished: assistive devices (a) that are not designed specifically for speech, (b) that take the average characteristics of speech into account, (c) that process articulatory or phonetic characteristics of speech, and (d) that embody some degree of automatic speech recognition. Assistive devices for people with speech and/or language disabilities typically involve some form of speech synthesis or symbol generation for severe forms of language disability. Speech synthesis is also used in text-to-speech systems for sightless persons. Other applications of assistive technology involving voice communication include voice control of wheelchairs and other devices for people with mobility disabilities. Images Fig. 4 PMID:7479816

Processing of Speech Signals for Physical and Sensory Disabilities

NASA Astrophysics Data System (ADS)

Levitt, Harry

1995-10-01

Assistive technology involving voice communication is used primarily by people who are deaf, hard of hearing, or who have speech and/or language disabilities. It is also used to a lesser extent by people with visual or motor disabilities. A very wide range of devices has been developed for people with hearing loss. These devices can be categorized not only by the modality of stimulation [i.e., auditory, visual, tactile, or direct electrical stimulation of the auditory nerve (auditory-neural)] but also in terms of the degree of speech processing that is used. At least four such categories can be distinguished: assistive devices (a) that are not designed specifically for speech, (b) that take the average characteristics of speech into account, (c) that process articulatory or phonetic characteristics of speech, and (d) that embody some degree of automatic speech recognition. Assistive devices for people with speech and/or language disabilities typically involve some form of speech synthesis or symbol generation for severe forms of language disability. Speech synthesis is also used in text-to-speech systems for sightless persons. Other applications of assistive technology involving voice communication include voice control of wheelchairs and other devices for people with mobility disabilities.

Voice technology and BBN

NASA Technical Reports Server (NTRS)

Wolf, Jared J.

1977-01-01

The following research was discussed: (1) speech signal processing; (2) automatic speech recognition; (3) continuous speech understanding; (4) speaker recognition; (5) speech compression; (6) subjective and objective evaluation of speech communication system; (7) measurement of the intelligibility and quality of speech when degraded by noise or other masking stimuli; (8) speech synthesis; (9) instructional aids for second-language learning and for training of the deaf; and (10) investigation of speech correlates of psychological stress. Experimental psychology, control systems, and human factors engineering, which are often relevant to the proper design and operation of speech systems are described.

USSR Report, Cybernetics Computers and Automation Technology

DTIC Science & Technology

1985-09-05

understand each other excellently, although in their speech they frequently omit, it would seem, needed words. However, the life experience of the...participants in a conversa- tion and their perception of voice intonations and gestures make it possible to fill in the missing elements of speech ...the Soviet Union. Comrade M. S. Gorbachev’s speech pointed out that microelectronics, computer technology, instrument building and the whole

Cloud-Based Speech Technology for Assistive Technology Applications (CloudCAST).

PubMed

Cunningham, Stuart; Green, Phil; Christensen, Heidi; Atria, José Joaquín; Coy, André; Malavasi, Massimiliano; Desideri, Lorenzo; Rudzicz, Frank

2017-01-01

The CloudCAST platform provides a series of speech recognition services that can be integrated into assistive technology applications. The platform and the services provided by the public API are described. Several exemplar applications have been developed to demonstrate the platform to potential developers and users.

Automatic speech recognition and training for severely dysarthric users of assistive technology: the STARDUST project.

PubMed

Parker, Mark; Cunningham, Stuart; Enderby, Pam; Hawley, Mark; Green, Phil

2006-01-01

The STARDUST project developed robust computer speech recognizers for use by eight people with severe dysarthria and concomitant physical disability to access assistive technologies. Independent computer speech recognizers trained with normal speech are of limited functional use by those with severe dysarthria due to limited and inconsistent proximity to "normal" articulatory patterns. Severe dysarthric output may also be characterized by a small mass of distinguishable phonetic tokens making the acoustic differentiation of target words difficult. Speaker dependent computer speech recognition using Hidden Markov Models was achieved by the identification of robust phonetic elements within the individual speaker output patterns. A new system of speech training using computer generated visual and auditory feedback reduced the inconsistent production of key phonetic tokens over time.

Difficulties in Automatic Speech Recognition of Dysarthric Speakers and Implications for Speech-Based Applications Used by the Elderly: A Literature Review

ERIC Educational Resources Information Center

Young, Victoria; Mihailidis, Alex

2010-01-01

Despite their growing presence in home computer applications and various telephony services, commercial automatic speech recognition technologies are still not easily employed by everyone; especially individuals with speech disorders. In addition, relatively little research has been conducted on automatic speech recognition performance with older…

Fault-Tolerant Software-Defined Radio on Manycore

NASA Technical Reports Server (NTRS)

Ricketts, Scott

2015-01-01

Software-defined radio (SDR) platforms generally rely on field-programmable gate arrays (FPGAs) and digital signal processors (DSPs), but such architectures require significant software development. In addition, application demands for radiation mitigation and fault tolerance exacerbate programming challenges. MaXentric Technologies, LLC, has developed a manycore-based SDR technology that provides 100 times the throughput of conventional radiationhardened general purpose processors. Manycore systems (30-100 cores and beyond) have the potential to provide high processing performance at error rates that are equivalent to current space-deployed uniprocessor systems. MaXentric's innovation is a highly flexible radio, providing over-the-air reconfiguration; adaptability; and uninterrupted, real-time, multimode operation. The technology is also compliant with NASA's Space Telecommunications Radio System (STRS) architecture. In addition to its many uses within NASA communications, the SDR can also serve as a highly programmable research-stage prototyping device for new waveforms and other communications technologies. It can also support noncommunication codes on its multicore processor, collocated with the communications workload-reducing the size, weight, and power of the overall system by aggregating processing jobs to a single board computer.

Baseband processor development/test performance for 30/20 GHz SS-TDMA communication system

NASA Technical Reports Server (NTRS)

Brown, L.; Sabourin, D.; Attwood, S.

1984-01-01

The baseband processor (BBP) development for the 30/20 GHz Satellite Communication System is described. The SS-TDMA concept for future satellite communications is reviewed, describing the overall system, the satellite payload, and the frequency plan. A brief general description of the BBP is given, and the proof-of-concept model of the BBP is summarized. Key technologies and custom LSI developed for the BBP are listed. Finally, key technology developments and test data are reported for the BBP.

Implicit, nonswitching, vector-oriented algorithm for steady transonic flow

NASA Technical Reports Server (NTRS)

Lottati, I.

1983-01-01

A rapid computation of a sequence of transonic flow solutions has to be performed in many areas of aerodynamic technology. The employment of low-cost vector array processors makes the conduction of such calculations economically feasible. However, for a full utilization of the new hardware, the developed algorithms must take advantage of the special characteristics of the vector array processor. The present investigation has the objective to develop an efficient algorithm for solving transonic flow problems governed by mixed partial differential equations on an array processor.

High Performance Processors for Space Environments: A Subproject of the NASA Exploration Missions Systems Directorate "Radiation Hardened Electronics for Space Environments" Technology Development Program

NASA Technical Reports Server (NTRS)

Johnson, M.; Label, K.; McCabe, J.; Powell, W.; Bolotin, G.; Kolawa, E.; Ng, T.; Hyde, D.

2007-01-01

Implementation of challenging Exploration Systems Missions Directorate objectives and strategies can be constrained by onboard computing capabilities and power efficiencies. The Radiation Hardened Electronics for Space Environments (RHESE) High Performance Processors for Space Environments project will address this challenge by significantly advancing the sustained throughput and processing efficiency of high-per$ormance radiation-hardened processors, targeting delivery of products by the end of FY12.

Backend Control Processor for a Multi-Processor Relational Database Computer System.

DTIC Science & Technology

1984-12-01

SCHOOL OF ENGI. UNCRSIFID MPONTIFF DEC 84 AFXT/GCS/ENG/84D-22 F/O 9/2 L ommhhhhmhhml mhhhommhhhhhm i-2 8 -- U0. 11111= Q. 2 111.8IIII- 1111111..6...THESIS Presented to the Faculty of the School of Engineering of the Air Force Institute of Technology Air University In Partial Fulfillment of the...development of a Backend Multi-Processor Relational Database Computer System. This thesis addresses a single component of this system, the Backend Control

Music Perception with Cochlear Implants: A Review

PubMed Central

McDermott, Hugh J.

2004-01-01

The acceptance of cochlear implantation as an effective and safe treatment for deafness has increased steadily over the past quarter century. The earliest devices were the first implanted prostheses found to be successful in compensating partially for lost sensory function by direct electrical stimulation of nerves. Initially, the main intention was to provide limited auditory sensations to people with profound or total sensorineural hearing impairment in both ears. Although the first cochlear implants aimed to provide patients with little more than awareness of environmental sounds and some cues to assist visual speech-reading, the technology has advanced rapidly. Currently, most people with modern cochlear implant systems can understand speech using the device alone, at least in favorable listening conditions. In recent years, an increasing research effort has been directed towards implant users’ perception of nonspeech sounds, especially music. This paper reviews that research, discusses the published experimental results in terms of both psychophysical observations and device function, and concludes with some practical suggestions about how perception of music might be enhanced for implant recipients in the future. The most significant findings of past research are: (1) On average, implant users perceive rhythm about as well as listeners with normal hearing; (2) Even with technically sophisticated multiple-channel sound processors, recognition of melodies, especially without rhythmic or verbal cues, is poor, with performance at little better than chance levels for many implant users; (3) Perception of timbre, which is usually evaluated by experimental procedures that require subjects to identify musical instrument sounds, is generally unsatisfactory; (4) Implant users tend to rate the quality of musical sounds as less pleasant than listeners with normal hearing; (5) Auditory training programs that have been devised specifically to provide implant users with structured musical listening experience may improve the subjective acceptability of music that is heard through a prosthesis; (6) Pitch perception might be improved by designing innovative sound processors that use both temporal and spatial patterns of electric stimulation more effectively and precisely to overcome the inherent limitations of signal coding in existing implant systems; (7) For the growing population of implant recipients who have usable acoustic hearing, at least for low-frequency sounds, perception of music is likely to be much better with combined acoustic and electric stimulation than is typical for deaf people who rely solely on the hearing provided by their prostheses. PMID:15497033

Music perception with cochlear implants: a review.

PubMed

McDermott, Hugh J

2004-01-01

The acceptance of cochlear implantation as an effective and safe treatment for deafness has increased steadily over the past quarter century. The earliest devices were the first implanted prostheses found to be successful in compensating partially for lost sensory function by direct electrical stimulation of nerves. Initially, the main intention was to provide limited auditory sensations to people with profound or total sensorineural hearing impairment in both ears. Although the first cochlear implants aimed to provide patients with little more than awareness of environmental sounds and some cues to assist visual speech-reading, the technology has advanced rapidly. Currently, most people with modern cochlear implant systems can understand speech using the device alone, at least in favorable listening conditions. In recent years, an increasing research effort has been directed towards implant users' perception of nonspeech sounds, especially music. This paper reviews that research, discusses the published experimental results in terms of both psychophysical observations and device function, and concludes with some practical suggestions about how perception of music might be enhanced for implant recipients in the future. The most significant findings of past research are: (1) On average, implant users perceive rhythm about as well as listeners with normal hearing; (2) Even with technically sophisticated multiple-channel sound processors, recognition of melodies, especially without rhythmic or verbal cues, is poor, with performance at little better than chance levels for many implant users; (3) Perception of timbre, which is usually evaluated by experimental procedures that require subjects to identify musical instrument sounds, is generally unsatisfactory; (4) Implant users tend to rate the quality of musical sounds as less pleasant than listeners with normal hearing; (5) Auditory training programs that have been devised specifically to provide implant users with structured musical listening experience may improve the subjective acceptability of music that is heard through a prosthesis; (6) Pitch perception might be improved by designing innovative sound processors that use both temporal and spatial patterns of electric stimulation more effectively and precisely to overcome the inherent limitations of signal coding in existing implant systems; (7) For the growing population of implant recipients who have usable acoustic hearing, at least for low-frequency sounds, perception of music is likely to be much better with combined acoustic and electric stimulation than is typical for deaf people who rely solely on the hearing provided by their prostheses.

Effects of Shared Active Surface Technology on the Communication and Speech of Two Preschool Children with Disabilities

ERIC Educational Resources Information Center

Travers, Jason C.; Fefer, Sarah A.

2017-01-01

Shared active surface (SAS) technology can be described as a supersized tablet computer for multiple simultaneous users. SAS technology has the potential to resolve issues historically associated with learning via single-user computer technology. This study reports findings of a SAS on the social communication and nonsocial speech of two preschool…

Experiences of the use of FOX, an intelligent agent, for programming cochlear implant sound processors in new users.

PubMed

Vaerenberg, Bart; Govaerts, Paul J; de Ceulaer, Geert; Daemers, Kristin; Schauwers, Karen

2011-01-01

This report describes the application of the software tool "Fitting to Outcomes eXpert" (FOX) in programming the cochlear implant (CI) processor in new users. FOX is an intelligent agent to assist in the programming of CI processors. The concept of FOX is to modify maps on the basis of specific outcome measures, achieved using heuristic logic and based on a set of deterministic "rules". A prospective study was conducted on eight consecutive CI-users with a follow-up of three months. Eight adult subjects with postlingual deafness were implanted with the Advanced Bionics HiRes90k device. The implants were programmed using FOX, running a set of rules known as Eargroup's EG0910 advice, which features a set of "automaps". The protocol employed for the initial 3 months is presented, with description of the map modifications generated by FOX and the corresponding psychoacoustic test results. The 3 month median results show 25 dBHL as PTA, 77% (55 dBSPL) and 71% (70 dBSPL) phoneme score at speech audiometry and loudness scaling in or near to the normal zone at different frequencies. It is concluded that this approach is feasible to start up CI fitting and yields good outcome.

Prediction of Pork Longissimus Lean Color Stability Using VIS/NIR

USDA-ARS?s Scientific Manuscript database

Insufficient case-life is a costly problem facing pork processors. To assess Visible and Near-Infrared (VIS/NIR) spectroscopy as a technology to sort pork loins according to lean color stability, center-cut pork loins (n = 1208) were selected from the boning lines of four large-scale pork processor...

Managing Power Heterogeneity

NASA Astrophysics Data System (ADS)

Pruhs, Kirk

A particularly important emergent technology is heterogeneous processors (or cores), which many computer architects believe will be the dominant architectural design in the future. The main advantage of a heterogeneous architecture, relative to an architecture of identical processors, is that it allows for the inclusion of processors whose design is specialized for particular types of jobs, and for jobs to be assigned to a processor best suited for that job. Most notably, it is envisioned that these heterogeneous architectures will consist of a small number of high-power high-performance processors for critical jobs, and a larger number of lower-power lower-performance processors for less critical jobs. Naturally, the lower-power processors would be more energy efficient in terms of the computation performed per unit of energy expended, and would generate less heat per unit of computation. For a given area and power budget, heterogeneous designs can give significantly better performance for standard workloads. Moreover, even processors that were designed to be homogeneous, are increasingly likely to be heterogeneous at run time: the dominant underlying cause is the increasing variability in the fabrication process as the feature size is scaled down (although run time faults will also play a role). Since manufacturing yields would be unacceptably low if every processor/core was required to be perfect, and since there would be significant performance loss from derating the entire chip to the functioning of the least functional processor (which is what would be required in order to attain processor homogeneity), some processor heterogeneity seems inevitable in chips with many processors/cores.

Robust Frequency Invariant Beamforming with Low Sidelobe for Speech Enhancement

NASA Astrophysics Data System (ADS)

Zhu, Yiting; Pan, Xiang

2018-01-01

Frequency invariant beamformers (FIBs) are widely used in speech enhancement and source localization. There are two traditional optimization methods for FIB design. The first one is convex optimization, which is simple but the frequency invariant characteristic of the beam pattern is poor with respect to frequency band of five octaves. The least squares (LS) approach using spatial response variation (SRV) constraint is another optimization method. Although, it can provide good frequency invariant property, it usually couldn’t be used in speech enhancement for its lack of weight norm constraint which is related to the robustness of a beamformer. In this paper, a robust wideband beamforming method with a constant beamwidth is proposed. The frequency invariant beam pattern is achieved by resolving an optimization problem of the SRV constraint to cover speech frequency band. With the control of sidelobe level, it is available for the frequency invariant beamformer (FIB) to prevent distortion of interference from the undesirable direction. The approach is completed in time-domain by placing tapped delay lines(TDL) and finite impulse response (FIR) filter at the output of each sensor which is more convenient than the Frost processor. By invoking the weight norm constraint, the robustness of the beamformer is further improved against random errors. Experiment results show that the proposed method has a constant beamwidth and almost the same white noise gain as traditional delay-and-sum (DAS) beamformer.

An Investigation of the Compensatory Effectiveness of Assistive Technology on Postsecondary Students with Learning Disabilities. Final Report.

ERIC Educational Resources Information Center

Murphy, Harry; Higgins, Eleanor

This final report describes the activities and accomplishments of a 3-year study on the compensatory effectiveness of three assistive technologies, optical character recognition, speech synthesis, and speech recognition, on postsecondary students (N=140) with learning disabilities. These technologies were investigated relative to: (1) immediate…

Teaching Speech Communication in a Black College: Does Technology Make a Difference?

ERIC Educational Resources Information Center

Nwadike, Fellina O.; Ekeanyanwu, Nnamdi T.

2011-01-01

Teaching a speech communication course in typical HBCUs (historically black colleges and universities) comes with many issues, because the application of technology in some minority institutions differs. The levels of acceptability as well as affordability are also core issues that affect application. Using technology in the classroom means many…

Status of a Power Processor for the Prometheus-1 Electric Propulsion System

NASA Technical Reports Server (NTRS)

Pinero, Luis R.; Hill, Gerald M.; Aulisio, Michael; Gerber, Scott; Griebeler, Elmer; Hewitt, Frank; Scina, Joseph

2006-01-01

NASA is developing technologies for nuclear electric propulsion for proposed deep space missions in support of the Exploration initiative under Project Prometheus. Electrical power produced by the combination of a fission-based power source and a Brayton power conversion and distribution system is used by a high specific impulse ion propulsion system to propel the spaceship. The ion propulsion system include the thruster, power processor and propellant feed system. A power processor technology development effort was initiated under Project Prometheus to develop high performance and lightweight power-processing technologies suitable for the application. This effort faces multiple challenges including developing radiation hardened power modules and converters with very high power capability and efficiency to minimize the impact on the power conversion and distribution system as well as the heat rejection system. This paper documents the design and test results of the first version of the beam supply, the design of a second version of the beam supply and the design and test results of the ancillary supplies.

Research in speech communication.

PubMed Central

Flanagan, J

1995-01-01

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker. Images Fig. 1 Fig. 2 Fig. 5 Fig. 8 Fig. 11 Fig. 12 Fig. 13 PMID:7479806

Lip Movement Exaggerations during Infant-Directed Speech

ERIC Educational Resources Information Center

Green, Jordan R.; Nip, Ignatius S. B.; Wilson, Erin M.; Mefferd, Antje S.; Yunusova, Yana

2010-01-01

Purpose: Although a growing body of literature has identified the positive effects of visual speech on speech and language learning, oral movements of infant-directed speech (IDS) have rarely been studied. This investigation used 3-dimensional motion capture technology to describe how mothers modify their lip movements when talking to their…

Automated Assessment of Speech Fluency for L2 English Learners

ERIC Educational Resources Information Center

Yoon, Su-Youn

2009-01-01

This dissertation provides an automated scoring method of speech fluency for second language learners of English (L2 learners) based that uses speech recognition technology. Non-standard pronunciation, frequent disfluencies, faulty grammar, and inappropriate lexical choices are crucial characteristics of L2 learners' speech. Due to the ease of…

Voice gender identification by cochlear implant users: The role of spectral and temporal resolution

NASA Astrophysics Data System (ADS)

Fu, Qian-Jie; Chinchilla, Sherol; Nogaki, Geraldine; Galvin, John J.

2005-09-01

The present study explored the relative contributions of spectral and temporal information to voice gender identification by cochlear implant users and normal-hearing subjects. Cochlear implant listeners were tested using their everyday speech processors, while normal-hearing subjects were tested under speech processing conditions that simulated various degrees of spectral resolution, temporal resolution, and spectral mismatch. Voice gender identification was tested for two talker sets. In Talker Set 1, the mean fundamental frequency values of the male and female talkers differed by 100 Hz while in Talker Set 2, the mean values differed by 10 Hz. Cochlear implant listeners achieved higher levels of performance with Talker Set 1, while performance was significantly reduced for Talker Set 2. For normal-hearing listeners, performance was significantly affected by the spectral resolution, for both Talker Sets. With matched speech, temporal cues contributed to voice gender identification only for Talker Set 1 while spectral mismatch significantly reduced performance for both Talker Sets. The performance of cochlear implant listeners was similar to that of normal-hearing subjects listening to 4-8 spectral channels. The results suggest that, because of the reduced spectral resolution, cochlear implant patients may attend strongly to periodicity cues to distinguish voice gender.

A wideband software reconfigurable modem

NASA Astrophysics Data System (ADS)

Turner, J. H., Jr.; Vickers, H.

A wideband modem is described which provides signal processing capability for four Lx-band signals employing QPSK, MSK and PPM waveforms and employs a software reconfigurable architecture for maximum system flexibility and graceful degradation. The current processor uses a 2901 and two 8086 microprocessors per channel and performs acquisition, tracking, and data demodulation for JITDS, GPS, IFF and TACAN systems. The next generation processor will be implemented using a VHSIC chip set employing a programmable complex array vector processor module, a GP computer module, customized gate array modules, and a digital array correlator. This integrated processor has application to a wide number of diverse system waveforms, and will bring the benefits of VHSIC technology insertion into avionic antijam communications systems.

Future technology in cochlear implants: assessing the benefit.

PubMed

Briggs, Robert J S

2011-05-01

It has been over 50 years since Djourno and Eyries first attempted electric stimulation in a patient with deafness. Over this time, the Cochlear Implant (CI) has become not only remarkably successful, but increasingly complex. Although the basic components of the system still comprise an implanted receiver stimulator and electrode, externally worn speech processor, microphone, control system, and power source, there are now several alternative designs of these components with different attributes that can be variably combined to meet the needs of specific patient groups. Development by the manufacturers has been driven both by these various patient needs, and also by the desire to achieve technological superiority, or at least differentiation, ultimately in pursuit of market share. Assessment of benefit is the responsibility of clinicians. It is incumbent on both industry and clinicians to ensure appropriate, safe, and affordable introduction of new technology. For example, experience with the totally implanted cochlear implant (TIKI) has demonstrated that quality of hearing is the over-riding consideration for CI users. To date, improved hearing outcomes have been achieved by improvements in: speech processing strategies; microphone technology; pre-processing strategies; electrode placement; bilateral implantation; use of a hearing aid in the opposite ear (bimodal stimulation); and the combination of electric and acoustic stimulation in the same ear. The resulting expansion of CI candidacy, with more residual hearing, further improves the outcomes achieved. Largely facilitated by advances in electronic capability and computerization, it can be expected that these improvements will continue. However, marked variability of results still occurs and we cannot assure any individual patient of their outcome. Realistic goals for implementation of new technology include: improved hearing in noise and music perception; effective invisible hearing (no external apparatus); automated fitting; and reduction in outcome variability. This paper provides examples of relevant potential future technologies that can be applied to reach these goals. In the quest for better outcomes, future technology must deliver improved reliability and usability for both clinicians and recipients that does not compromise safety and is affordable. One of the challenges related to the introduction of new technologies is the 'classification' of CI systems and the framework under which sufficient change and increased benefit can be demonstrated to establish a claim of 'new generation CI' and hence increased reimbursement from third-party payers. Significant improvements in hearing outcomes and quality of life associated with CI design changes are difficult to measure, particularly when there is such dramatic benefit from the intervention of cochlear implantation from the individual's perspective. Manufacturers and clinicians need to be objective and undertake appropriate safety studies and long-term and multi-centre clinical trials to ensure that the introduction of new technology is both safe and effective and supported by health systems worldwide.

Matching Automatic Gain Control Across Devices in Bimodal Cochlear Implant Users.

PubMed

Veugen, Lidwien C E; Chalupper, Josef; Snik, Ad F M; Opstal, A John van; Mens, Lucas H M

2016-01-01

The purpose of this study was to improve bimodal benefit in listeners using a cochlear implant (CI) and a hearing aid (HA) in contralateral ears, by matching the time constants and the number of compression channels of the automatic gain control (AGC) of the HA to the CI. Equivalent AGC was hypothesized to support a balanced loudness for dynamically changing signals like speech and improve bimodal benefit for speech understanding in quiet and with noise presented from the side(s) at 90 degree. Fifteen subjects participated in the study, all using the same Advanced Bionics Harmony CI processor and HA (Phonak Naida S IX UP). In a 3-visit crossover design with 4 weeks between sessions, performance was measured using a HA with a standard AGC (syllabic multichannel compression with 1 ms attack time and 50 ms release time) or an AGC that was adjusted to match that of the CI processor (dual AGC broadband compression, 3 and 240 msec attack time, 80 and 1500 msec release time). In all devices, the AGC was activated above the threshold of 63 dB SPL. The authors balanced loudness across the devices for soft and loud input sounds in 3 frequency bands (0 to 548, 548 to 1000, and >1000 Hz). Speech understanding was tested in free field in quiet and in noise for three spatial speaker configurations, with target speech always presented from the front. Single-talker noise was either presented from the CI side or the HA side, or uncorrelated stationary speech-weighted noise or single-talker noise was presented from both sides. Questionnaires were administered to assess differences in perception between the two bimodal fittings. Significant bimodal benefit over the CI alone was only found for the AGC-matched HA for the speech tests with single-talker noise. Compared with the standard HA, matched AGC characteristics significantly improved speech understanding in single-talker noise by 1.9 dB when noise was presented from the HA side. AGC matching increased bimodal benefit insignificantly by 0.6 dB when noise was presented from the CI implanted side, or by 0.8 (single-talker noise) and 1.1 dB (stationary noise) in the more complex configurations with two simultaneous maskers from both sides. In questionnaires, subjects rated the AGC-matched HA higher than the standard HA for understanding of one person in quiet and in noise, and for the quality of sounds. Listening to a slightly raised voice, subjects indicated increased listening comfort with matched AGCs. At the end of the study, 9 of 15 subjects preferred to take home the AGC-matched HA, 1 preferred the standard HA and 5 subjects had no preference. For bimodal listening, the AGC-matched HA outperformed the standard HA in speech understanding in noise tasks using a single competing talker and it was favored in questionnaires and in a subjective preference test. When noise was presented from the HA side, AGC matching resulted in a 1.9 dB SNR additional benefit, even though the HA was at the least favorable SNR side in this speaker configuration. Our results possibly suggest better binaural processing for matched AGCs.

The Affordance of Speech Recognition Technology for EFL Learning in an Elementary School Setting

ERIC Educational Resources Information Center

Liaw, Meei-Ling

2014-01-01

This study examined the use of speech recognition (SR) technology to support a group of elementary school children's learning of English as a foreign language (EFL). SR technology has been used in various language learning contexts. Its application to EFL teaching and learning is still relatively recent, but a solid understanding of its…

The Effect of Automatic Speech Recognition Eyespeak Software on Iraqi Students' English Pronunciation: A Pilot Study

ERIC Educational Resources Information Center

Sidgi, Lina Fathi Sidig; Shaari, Ahmad Jelani

2017-01-01

The use of technology, such as computer-assisted language learning (CALL), is used in teaching and learning in the foreign language classrooms where it is most needed. One promising emerging technology that supports language learning is automatic speech recognition (ASR). Integrating such technology, especially in the instruction of pronunciation…

Yaounde French Speech Corpus

DTIC Science & Technology

2017-03-01

the Center for Technology Enhanced Language Learning (CTELL), a research cell in the Department of Foreign Languages, United States Military Academy...models for automatic speech recognition (ASR), and to, thereby, investigate the utility of ASR in pedagogical technology . The corpus is a sample of...lexical resources, language technology 16. SECURITY CLASSIFICATION OF: 17. LIMITATION OF ABSTRACT UU 18. NUMBER OF

Advances in real-time magnetic resonance imaging of the vocal tract for speech science and technology research.

PubMed

Toutios, Asterios; Narayanan, Shrikanth S

2016-01-01

Real-time magnetic resonance imaging (rtMRI) of the moving vocal tract during running speech production is an important emerging tool for speech production research providing dynamic information of a speaker's upper airway from the entire mid-sagittal plane or any other scan plane of interest. There have been several advances in the development of speech rtMRI and corresponding analysis tools, and their application to domains such as phonetics and phonological theory, articulatory modeling, and speaker characterization. An important recent development has been the open release of a database that includes speech rtMRI data from five male and five female speakers of American English each producing 460 phonetically balanced sentences. The purpose of the present paper is to give an overview and outlook of the advances in rtMRI as a tool for speech research and technology development.

Advances in real-time magnetic resonance imaging of the vocal tract for speech science and technology research

PubMed Central

TOUTIOS, ASTERIOS; NARAYANAN, SHRIKANTH S.

2016-01-01

Real-time magnetic resonance imaging (rtMRI) of the moving vocal tract during running speech production is an important emerging tool for speech production research providing dynamic information of a speaker's upper airway from the entire mid-sagittal plane or any other scan plane of interest. There have been several advances in the development of speech rtMRI and corresponding analysis tools, and their application to domains such as phonetics and phonological theory, articulatory modeling, and speaker characterization. An important recent development has been the open release of a database that includes speech rtMRI data from five male and five female speakers of American English each producing 460 phonetically balanced sentences. The purpose of the present paper is to give an overview and outlook of the advances in rtMRI as a tool for speech research and technology development. PMID:27833745

Three Trailblazing Technologies for Schools.

ERIC Educational Resources Information Center

McGinty, Tony

1987-01-01

Provides an overview of the capabilities and potential educational applications of CD-ROM (compact disk read-only memory), artificial intelligence, and speech technology. Highlights include reference materials on CD-ROM; current developments in CD-I (compact disk interactive); synthesized and digital speech for microcomputers, including specific…

Earth Orbiter 1 (EO-1): Wideband Advanced Recorder and Processor (WARP)

NASA Technical Reports Server (NTRS)

Smith, Terry; Kessler, John

1999-01-01

An overview of the Earth Orbitor 1 (EO1) Wideband Advanced Recorder and Processor (WARP) is presented in viewgraph form. The WARP is a spacecraft component that receives, stores, and processes high rate science data and its associated ancillary data from multispectral detectors, hyperspectral detectors, and an atmospheric corrector, and then transmits the data via an X-band or S-band transmitter to the ground station. The WARP project goals are: (1) Pathfinder for next generation LANDSAT mission; (2) Flight prove architectures and technologies; and (3) Identify future technology needs.

NSTAR Ion Thrusters and Power Processors

NASA Technical Reports Server (NTRS)

Bond, T. A.; Christensen, J. A.

1999-01-01

The purpose of the NASA Solar Electric Propulsion Technology Applications Readiness (NSTAR) project is to validate ion propulsion technology for use on future NASA deep space missions. This program, which was initiated in September 1995, focused on the development of two sets of flight quality ion thrusters, power processors, and controllers that provided the same performance as engineering model hardware and also met the dynamic and environmental requirements of the Deep Space 1 Project. One of the flight sets was used for primary propulsion for the Deep Space 1 spacecraft which was launched in October 1998.

Development Of A Three-Dimensional Circuit Integration Technology And Computer Architecture

NASA Astrophysics Data System (ADS)

Etchells, R. D.; Grinberg, J.; Nudd, G. R.

1981-12-01

This paper is the first of a series 1,2,3 describing a range of efforts at Hughes Research Laboratories, which are collectively referred to as "Three-Dimensional Microelectronics." The technology being developed is a combination of a unique circuit fabrication/packaging technology and a novel processing architecture. The packaging technology greatly reduces the parasitic impedances associated with signal-routing in complex VLSI structures, while simultaneously allowing circuit densities orders of magnitude higher than the current state-of-the-art. When combined with the 3-D processor architecture, the resulting machine exhibits a one- to two-order of magnitude simultaneous improvement over current state-of-the-art machines in the three areas of processing speed, power consumption, and physical volume. The 3-D architecture is essentially that commonly referred to as a "cellular array", with the ultimate implementation having as many as 512 x 512 processors working in parallel. The three-dimensional nature of the assembled machine arises from the fact that the chips containing the active circuitry of the processor are stacked on top of each other. In this structure, electrical signals are passed vertically through the chips via thermomigrated aluminum feedthroughs. Signals are passed between adjacent chips by micro-interconnects. This discussion presents a broad view of the total effort, as well as a more detailed treatment of the fabrication and packaging technologies themselves. The results of performance simulations of the completed 3-D processor executing a variety of algorithms are also presented. Of particular pertinence to the interests of the focal-plane array community is the simulation of the UNICORNS nonuniformity correction algorithms as executed by the 3-D architecture.

The benefits of remote microphone technology for adults with cochlear implants.

PubMed

Fitzpatrick, Elizabeth M; Séguin, Christiane; Schramm, David R; Armstrong, Shelly; Chénier, Josée

2009-10-01

Cochlear implantation has become a standard practice for adults with severe to profound hearing loss who demonstrate limited benefit from hearing aids. Despite the substantial auditory benefits provided by cochlear implants, many adults experience difficulty understanding speech in noisy environments and in other challenging listening conditions such as television. Remote microphone technology may provide some benefit in these situations; however, little is known about whether these systems are effective in improving speech understanding in difficult acoustic environments for this population. This study was undertaken with adult cochlear implant recipients to assess the potential benefits of remote microphone technology. The objectives were to examine the measurable and perceived benefit of remote microphone devices during television viewing and to assess the benefits of a frequency-modulated system for speech understanding in noise. Fifteen adult unilateral cochlear implant users were fit with remote microphone devices in a clinical environment. The study used a combination of direct measurements and patient perceptions to assess speech understanding with and without remote microphone technology. The direct measures involved a within-subject repeated-measures design. Direct measures of patients' speech understanding during television viewing were collected using their cochlear implant alone and with their implant device coupled to an assistive listening device. Questionnaires were administered to document patients' perceptions of benefits during the television-listening tasks. Speech recognition tests of open-set sentences in noise with and without remote microphone technology were also administered. Participants showed improved speech understanding for television listening when using remote microphone devices coupled to their cochlear implant compared with a cochlear implant alone. This benefit was documented both when listening to news and talk show recordings. Questionnaire results also showed statistically significant differences between listening with a cochlear implant alone and listening with a remote microphone device. Participants judged that remote microphone technology provided them with better comprehension, more confidence, and greater ease of listening. Use of a frequency-modulated system coupled to a cochlear implant also showed significant improvement over a cochlear implant alone for open-set sentence recognition in +10 and +5 dB signal to noise ratios. Benefits were measured during remote microphone use in focused-listening situations in a clinical setting, for both television viewing and speech understanding in noise in the audiometric sound suite. The results suggest that adult cochlear implant users should be counseled regarding the potential for enhanced speech understanding in difficult listening environments through the use of remote microphone technology.

Performance of a low data rate speech codec for land-mobile satellite communications

NASA Technical Reports Server (NTRS)

Gersho, Allen; Jedrey, Thomas C.

1990-01-01

In an effort to foster the development of new technologies for the emerging land mobile satellite communications services, JPL funded two development contracts in 1984: one to the Univ. of Calif., Santa Barbara and the other to the Georgia Inst. of Technology, to develop algorithms and real time hardware for near toll quality speech compression at 4800 bits per second. Both universities have developed and delivered speech codecs to JPL, and the UCSB codec was extensively tested by JPL in a variety of experimental setups. The basic UCSB speech codec algorithms and the test results of the various experiments performed with this codec are presented.

Feasibility of automated speech sample collection with stuttering children using interactive voice response (IVR) technology.

PubMed

Vogel, Adam P; Block, Susan; Kefalianos, Elaina; Onslow, Mark; Eadie, Patricia; Barth, Ben; Conway, Laura; Mundt, James C; Reilly, Sheena

2015-04-01

To investigate the feasibility of adopting automated interactive voice response (IVR) technology for remotely capturing standardized speech samples from stuttering children. Participants were 10 6-year-old stuttering children. Their parents called a toll-free number from their homes and were prompted to elicit speech from their children using a standard protocol involving conversation, picture description and games. The automated IVR system was implemented using an off-the-shelf telephony software program and delivered by a standard desktop computer. The software infrastructure utilizes voice over internet protocol. Speech samples were automatically recorded during the calls. Video recordings were simultaneously acquired in the home at the time of the call to evaluate the fidelity of the telephone collected samples. Key outcome measures included syllables spoken, percentage of syllables stuttered and an overall rating of stuttering severity using a 10-point scale. Data revealed a high level of relative reliability in terms of intra-class correlation between the video and telephone acquired samples on all outcome measures during the conversation task. Findings were less consistent for speech samples during picture description and games. Results suggest that IVR technology can be used successfully to automate remote capture of child speech samples.

[Research on Barrier-free Home Environment System Based on Speech Recognition].

PubMed

Zhu, Husheng; Yu, Hongliu; Shi, Ping; Fang, Youfang; Jian, Zhuo

2015-10-01

The number of people with physical disabilities is increasing year by year, and the trend of population aging is more and more serious. In order to improve the quality of the life, a control system of accessible home environment for the patients with serious disabilities was developed to control the home electrical devices with the voice of the patients. The control system includes a central control platform, a speech recognition module, a terminal operation module, etc. The system combines the speech recognition control technology and wireless information transmission technology with the embedded mobile computing technology, and interconnects the lamp, electronic locks, alarms, TV and other electrical devices in the home environment as a whole system through a wireless network node. The experimental results showed that speech recognition success rate was more than 84% in the home environment.

Design of a dataway processor for a parallel image signal processing system

NASA Astrophysics Data System (ADS)

Nomura, Mitsuru; Fujii, Tetsuro; Ono, Sadayasu

1995-04-01

Recently, demands for high-speed signal processing have been increasing especially in the field of image data compression, computer graphics, and medical imaging. To achieve sufficient power for real-time image processing, we have been developing parallel signal-processing systems. This paper describes a communication processor called 'dataway processor' designed for a new scalable parallel signal-processing system. The processor has six high-speed communication links (Dataways), a data-packet routing controller, a RISC CORE, and a DMA controller. Each communication link operates at 8-bit parallel in a full duplex mode at 50 MHz. Moreover, data routing, DMA, and CORE operations are processed in parallel. Therefore, sufficient throughput is available for high-speed digital video signals. The processor is designed in a top- down fashion using a CAD system called 'PARTHENON.' The hardware is fabricated using 0.5-micrometers CMOS technology, and its hardware is about 200 K gates.

Parallel volume ray-casting for unstructured-grid data on distributed-memory architectures

NASA Technical Reports Server (NTRS)

Ma, Kwan-Liu

1995-01-01

As computing technology continues to advance, computational modeling of scientific and engineering problems produces data of increasing complexity: large in size and unstructured in shape. Volume visualization of such data is a challenging problem. This paper proposes a distributed parallel solution that makes ray-casting volume rendering of unstructured-grid data practical. Both the data and the rendering process are distributed among processors. At each processor, ray-casting of local data is performed independent of the other processors. The global image composing processes, which require inter-processor communication, are overlapped with the local ray-casting processes to achieve maximum parallel efficiency. This algorithm differs from previous ones in four ways: it is completely distributed, less view-dependent, reasonably scalable, and flexible. Without using dynamic load balancing, test results on the Intel Paragon using from two to 128 processors show, on average, about 60% parallel efficiency.

Six characteristics of effective structured reporting and the inevitable integration with speech recognition.

PubMed

Liu, David; Zucherman, Mark; Tulloss, William B

2006-03-01

The reporting of radiological images is undergoing dramatic changes due to the introduction of two new technologies: structured reporting and speech recognition. Each technology has its own unique advantages. The highly organized content of structured reporting facilitates data mining and billing, whereas speech recognition offers a natural succession from the traditional dictation-transcription process. This article clarifies the distinction between the process and outcome of structured reporting, describes fundamental requirements for any effective structured reporting system, and describes the potential development of a novel, easy-to-use, customizable structured reporting system that incorporates speech recognition. This system should have all the advantages derived from structured reporting, accommodate a wide variety of user needs, and incorporate speech recognition as a natural component and extension of the overall reporting process.

Automated speech recognition for time recording in out-of-hospital emergency medicine-an experimental approach.

PubMed

Gröschel, J; Philipp, F; Skonetzki, St; Genzwürker, H; Wetter, Th; Ellinger, K

2004-02-01

Precise documentation of medical treatment in emergency medical missions and for resuscitation is essential from a medical, legal and quality assurance point of view [Anästhesiologie und Intensivmedizin, 41 (2000) 737]. All conventional methods of time recording are either too inaccurate or elaborate for routine application. Automated speech recognition may offer a solution. A special erase programme for the documentation of all time events was developed. Standard speech recognition software (IBM ViaVoice 7.0) was adapted and installed on two different computer systems. One was a stationary PC (500MHz Pentium III, 128MB RAM, Soundblaster PCI 128 Soundcard, Win NT 4.0), the other was a mobile pen-PC that had already proven its value during emergency missions [Der Notarzt 16, p. 177] (Fujitsu Stylistic 2300, 230Mhz MMX Processor, 160MB RAM, embedded soundcard ESS 1879 chipset, Win98 2nd ed.). On both computers two different microphones were tested. One was a standard headset that came with the recognition software, the other was a small microphone (Lavalier-Kondensatormikrofon EM 116 from Vivanco), that could be attached to the operators collar. Seven women and 15 men spoke a text with 29 phrases to be recognised. Two emergency physicians tested the system in a simulated emergency setting using the collar microphone and the pen-PC with an analogue wireless connection. Overall recognition was best for the PC with a headset (89%) followed by the pen-PC with a headset (85%), the PC with a microphone (84%) and the pen-PC with a microphone (80%). Nevertheless, the difference was not statistically significant. Recognition became significantly worse (89.5% versus 82.3%, P<0.0001 ) when numbers had to be recognised. The gender of speaker and the number of words in a sentence had no influence. Average recognition in the simulated emergency setting was 75%. At no time did false recognition appear. Time recording with automated speech recognition seems to be possible in emergency medical missions. Although results show an average recognition of only 75%, it is possible that missing elements may be reconstructed more precisely. Future technology should integrate a secure wireless connection between microphone and mobile computer. The system could then prove its value for real out-of-hospital emergencies.

Assistive Technology and Adults with Learning Disabilities: A Blueprint for Exploration and Advancement.

ERIC Educational Resources Information Center

Raskind, Marshall

1993-01-01

This article describes assistive technologies for persons with learning disabilities, including word processing, spell checking, proofreading programs, outlining/"brainstorming" programs, abbreviation expanders, speech recognition, speech synthesis/screen review, optical character recognition systems, personal data managers, free-form databases,…

Review of Speech-to-Text Recognition Technology for Enhancing Learning

ERIC Educational Resources Information Center

Shadiev, Rustam; Hwang, Wu-Yuin; Chen, Nian-Shing; Huang, Yueh-Min

2014-01-01

This paper reviewed literature from 1999 to 2014 inclusively on how Speech-to-Text Recognition (STR) technology has been applied to enhance learning. The first aim of this review is to understand how STR technology has been used to support learning over the past fifteen years, and the second is to analyze all research evidence to understand how…

Performance Evaluation of Speech Recognition Systems as a Next-Generation Pilot-Vehicle Interface Technology

NASA Technical Reports Server (NTRS)

Arthur, Jarvis J., III; Shelton, Kevin J.; Prinzel, Lawrence J., III; Bailey, Randall E.

2016-01-01

During the flight trials known as Gulfstream-V Synthetic Vision Systems Integrated Technology Evaluation (GV-SITE), a Speech Recognition System (SRS) was used by the evaluation pilots. The SRS system was intended to be an intuitive interface for display control (rather than knobs, buttons, etc.). This paper describes the performance of the current "state of the art" Speech Recognition System (SRS). The commercially available technology was evaluated as an application for possible inclusion in commercial aircraft flight decks as a crew-to-vehicle interface. Specifically, the technology is to be used as an interface from aircrew to the onboard displays, controls, and flight management tasks. A flight test of a SRS as well as a laboratory test was conducted.

Automatic speech recognition technology development at ITT Defense Communications Division

NASA Technical Reports Server (NTRS)

White, George M.

1977-01-01

An assessment of the applications of automatic speech recognition to defense communication systems is presented. Future research efforts include investigations into the following areas: (1) dynamic programming; (2) recognition of speech degraded by noise; (3) speaker independent recognition; (4) large vocabulary recognition; (5) word spotting and continuous speech recognition; and (6) isolated word recognition.

Speech Perception Benefits of FM and Infrared Devices to Children with Hearing Aids in a Typical Classroom

ERIC Educational Resources Information Center

Anderson, Karen L.; Goldstein, Howard

2004-01-01

Children typically learn in classroom environments that have background noise and reverberation that interfere with accurate speech perception. Amplification technology can enhance the speech perception of students who are hard of hearing. Purpose: This study used a single-subject alternating treatments design to compare the speech recognition…

Communications Processors: Categories, Applications, and Trends

DTIC Science & Technology

1976-03-01

allow switching from BSC to SDLC .(12) Standard protocols would ease the requirement that communications processor software convert from one...COMMANDER c^/g^_ (^-»M-^ V »*-^ FRANK J. EMMA, Colonel, USAF Director, information Systems Technology Applications Office Deputy for Command...guidelines in selecting a device for a specific application are included, with manufacturer models presented as illustrations. UNCLASSIFIED SECURITY

Cooperative use of advanced scanning technology for low-volume hardwood processors

Treesearch

Luis G. Occeña; Timothy J. Rayner; Daniel L. Schmoldt; A. Lynn Abbott

2001-01-01

Of the several hundreds of hardwood lumber sawmills across the country, the majority are small- to medium-sized facilities operated as small businesses in rural communities. Trends of increased log costs and limited availability are forcing wood processors to become more efficient in their operations. Still, small mills are less able to adopt new, more efficient...

Closing the Gap: Cybersecurity for U.S. Forces and Commands

DTIC Science & Technology

2017-03-30

Dickson, Ph.D. Professor of Military Studies , JAWS Thesis Advisor Kevin Therrien, Col, USAF Committee Member Stephen Rogers, Colonel, USA Director...infrastructures, and includes the Internet, telecommunications networks, computer systems, and embedded processors and controllers in critical industries.”5...of information technology infrastructures, including the Internet, telecommunications networks, computer systems, and embedded processors and

Custom large scale integrated circuits for spaceborne SAR processors

NASA Technical Reports Server (NTRS)

Tyree, V. C.

1978-01-01

The application of modern LSI technology to the development of a time-domain azimuth correlator for SAR processing is discussed. General design requirements for azimuth correlators for missions such as SEASAT-A, Venus orbital imaging radar (VOIR), and shuttle imaging radar (SIR) are summarized. Several azimuth correlator architectures that are suitable for implementation using custom LSI devices are described. Technical factors pertaining to selection of appropriate LSI technologies are discussed, and the maturity of alternative technologies for spacecraft applications are reported in the context of expected space mission launch dates. The preliminary design of a custom LSI time-domain azimuth correlator device (ACD) being developed for use in future SAR processors is detailed.

Performance Qualification Test of the ISS Water Processor Assembly (WPA) Expendables

NASA Technical Reports Server (NTRS)

Carter, Layne; Tabb, David; Tatara, James D.; Mason, Richard K.

2005-01-01

The Water Processor Assembly (WPA) for use on the International Space Station (ISS) includes various technologies for the treatment of waste water. These technologies include filtration, ion exchange, adsorption, catalytic oxidation, and iodination. The WPA hardware implementing portions of these technologies, including the Particulate Filter, Multifiltration Bed, Ion Exchange Bed, and Microbial Check Valve, was recently qualified for chemical performance at the Marshall Space Flight Center. Waste water representing the quality of that produced on the ISS was generated by test subjects and processed by the WPA. Water quality analysis and instrumentation data was acquired throughout the test to monitor hardware performance. This paper documents operation of the test and the assessment of the hardware performance.

Application of speech recognition and synthesis in the general aviation cockpit

NASA Technical Reports Server (NTRS)

North, R. A.; Mountford, S. J.; Bergeron, H.

1984-01-01

Interactive speech recognition/synthesis technology is assessed as a method for the aleviation of single-pilot IFR flight workloads. Attention was given during this series of evaluations to the conditions typical of general aviation twin-engine aircrft cockpits, covering several commonly encountered IFR flight condition scenarios. The most beneficial speech command tasks are noted to be in the data retrieval domain, which would allow the pilot access to uplinked data, checklists, and performance charts. Data entry tasks also appear to benefit from this technology.

Development of a Next-Generation Membrane-Integrated Adsorption Processor for CO2 Removal and Compression for Closed-Loop Air Revitalization Systems

NASA Technical Reports Server (NTRS)

Mulloth, Lila; LeVan, Douglas

2002-01-01

The current CO2 removal technology of NASA is very energy intensive and contains many non-optimized subsystems. This paper discusses the concept of a next-generation, membrane integrated, adsorption processor for CO2 removal nd compression in closed-loop air revitalization systems. This processor will use many times less power than NASA's current CO2 removal technology and will be capable of maintaining a lower CO2 concentration in the cabin than that can be achieved by the existing CO2 removal systems. The compact, consolidated, configuration of gas dryer, CO2 separator, and CO2 compressor will allow continuous recycling of humid air in the cabin and supply of compressed CO2 to the reduction unit for oxygen recovery. The device has potential application to the International Space Station and future, long duration, transit, and planetary missions.

SPEECH--MAN'S NATURAL COMMUNICATION.

ERIC Educational Resources Information Center

DUDLEY, HOMER; AND OTHERS

SESSION 63 OF THE 1967 INSTITUTE OF ELECTRICAL AND ELECTRONIC ENGINEERS INTERNATIONAL CONVENTION BROUGHT TOGETHER SEVEN DISTINGUISHED MEN WORKING IN FIELDS RELEVANT TO LANGUAGE. THEIR TOPICS INCLUDED ORIGIN AND EVOLUTION OF SPEECH AND LANGUAGE, LANGUAGE AND CULTURE, MAN'S PHYSIOLOGICAL MECHANISMS FOR SPEECH, LINGUISTICS, AND TECHNOLOGY AND…

An implementation of a reference symbol approach to generic modulation in fading channels

NASA Technical Reports Server (NTRS)

Young, R. J.; Lodge, J. H.; Pacola, L. C.

1990-01-01

As mobile satellite communications systems evolve over the next decade, they will have to adapt to a changing tradeoff between bandwidth and power. This paper presents a flexible approach to digital modulation and coding that will accommodate both wideband and narrowband schemes. This architecture could be the basis for a family of modems, each satisfying a specific power and bandwidth constraint, yet all having a large number of common signal processing blocks. The implementation of this generic approach, with general purpose digital processors for transmission of 4.8 kilobits per sec. digitally encoded speech, is described.

Augmentative and Alternative Communication in Autism: A Comparison of the Picture Exchange Communication System and Speech-Output Technology

ERIC Educational Resources Information Center

Boesch, Miriam Chacon

2011-01-01

The purpose of this comparative efficacy study was to investigate the Picture Exchange Communication System (PECS) and a speech-generating device (SGD) in developing requesting skills, social-communicative behavior, and speech for three elementary-age children with severe autism and little to no functional speech. Requesting was selected as the…

Using Automatic Speech Recognition to Dictate Mathematical Expressions: The Development of the "TalkMaths" Application at Kingston University

ERIC Educational Resources Information Center

Wigmore, Angela; Hunter, Gordon; Pflugel, Eckhard; Denholm-Price, James; Binelli, Vincent

2009-01-01

Speech technology--especially automatic speech recognition--has now advanced to a level where it can be of great benefit both to able-bodied people and those with various disabilities. In this paper we describe an application "TalkMaths" which, using the output from a commonly-used conventional automatic speech recognition system,…

A Prospectus for the Future Development of a Speech Lab: Hypertext Applications.

ERIC Educational Resources Information Center

Berube, David M.

This paper presents a plan for the next generation of speech laboratories which integrates technologies of modern communication in order to improve and modernize the instructional process. The paper first examines the application of intermediate technologies including audio-video recording and playback, computer assisted instruction and testing…

Voice Technologies in Libraries: A Look into the Future.

ERIC Educational Resources Information Center

Lange, Holley R., Ed.; And Others

1991-01-01

Discussion of synthesized speech and voice recognition focuses on a forum that addressed the potential for speech technologies in libraries. Topics discussed by three contributors include possible library applications in technical processing, book receipt, circulation control, and database access; use by disabled and illiterate users; and problems…

Using Computer Technology To Monitor Student Progress and Remediate Reading Problems.

ERIC Educational Resources Information Center

McCullough, C. Sue

1995-01-01

Focuses on research about application of text-to-speech systems in diagnosing and remediating word recognition, vocabulary knowledge, and comprehension disabilities. As school psychologists move toward a consultative model of service delivery, they need to know about technology such as speech synthesizers, digitizers, optical-character-recognition…

Fifty years of progress in speech and speaker recognition

NASA Astrophysics Data System (ADS)

Furui, Sadaoki

2004-10-01

Speech and speaker recognition technology has made very significant progress in the past 50 years. The progress can be summarized by the following changes: (1) from template matching to corpus-base statistical modeling, e.g., HMM and n-grams, (2) from filter bank/spectral resonance to Cepstral features (Cepstrum + DCepstrum + DDCepstrum), (3) from heuristic time-normalization to DTW/DP matching, (4) from gdistanceh-based to likelihood-based methods, (5) from maximum likelihood to discriminative approach, e.g., MCE/GPD and MMI, (6) from isolated word to continuous speech recognition, (7) from small vocabulary to large vocabulary recognition, (8) from context-independent units to context-dependent units for recognition, (9) from clean speech to noisy/telephone speech recognition, (10) from single speaker to speaker-independent/adaptive recognition, (11) from monologue to dialogue/conversation recognition, (12) from read speech to spontaneous speech recognition, (13) from recognition to understanding, (14) from single-modality (audio signal only) to multi-modal (audio/visual) speech recognition, (15) from hardware recognizer to software recognizer, and (16) from no commercial application to many practical commercial applications. Most of these advances have taken place in both the fields of speech recognition and speaker recognition. The majority of technological changes have been directed toward the purpose of increasing robustness of recognition, including many other additional important techniques not noted above.

A self-sustained, complete and miniaturized methanol fuel processor for proton exchange membrane fuel cell

NASA Astrophysics Data System (ADS)

Yang, Mei; Jiao, Fengjun; Li, Shulian; Li, Hengqiang; Chen, Guangwen

2015-08-01

A self-sustained, complete and miniaturized methanol fuel processor has been developed based on modular integration and microreactor technology. The fuel processor is comprised of one methanol oxidative reformer, one methanol combustor and one two-stage CO preferential oxidation unit. Microchannel heat exchanger is employed to recover heat from hot stream, miniaturize system size and thus achieve high energy utilization efficiency. By optimized thermal management and proper operation parameter control, the fuel processor can start up in 10 min at room temperature without external heating. A self-sustained state is achieved with H2 production rate of 0.99 Nm3 h-1 and extremely low CO content below 25 ppm. This amount of H2 is sufficient to supply a 1 kWe proton exchange membrane fuel cell. The corresponding thermal efficiency of whole processor is higher than 86%. The size and weight of the assembled reactors integrated with microchannel heat exchangers are 1.4 L and 5.3 kg, respectively, demonstrating a very compact construction of the fuel processor.

30/20 GHz communications systems baseband processor development

NASA Astrophysics Data System (ADS)

Brown, L.; Sabourin, D.; Stilwell, J.; McCallister, R.; Borota, M.

The architecture and system design concepts for a commercial satellite communications system planned for the 1990's has been developed. The system provides data communications between the individual users via trunking and customer premise service terminals utilizing a central switching satellite operating in a time-division multiple-access mode. Baseband processing is employed to route and control traffic on an individual message basis while providing significant advantages in improved link margins and system flexibility. Key technology developments required to prove the flight readiness of the baseband processor design are being verified in the baseband processor proof-of-concept model described herein.

30/20 GHz communications systems baseband processor development

NASA Technical Reports Server (NTRS)

Brown, L.; Sabourin, D.; Stilwell, J.; Mccallister, R.; Borota, M.

1982-01-01

The architecture and system design concepts for a commercial satellite communications system planned for the 1990's has been developed. The system provides data communications between the individual users via trunking and customer premise service terminals utilizing a central switching satellite operating in a time-division multiple-access mode. Baseband processing is employed to route and control traffic on an individual message basis while providing significant advantages in improved link margins and system flexibility. Key technology developments required to prove the flight readiness of the baseband processor design are being verified in the baseband processor proof-of-concept model described herein.

Positron Emission Tomography Imaging Reveals Auditory and Frontal Cortical Regions Involved with Speech Perception and Loudness Adaptation.

PubMed

Berding, Georg; Wilke, Florian; Rode, Thilo; Haense, Cathleen; Joseph, Gert; Meyer, Geerd J; Mamach, Martin; Lenarz, Minoo; Geworski, Lilli; Bengel, Frank M; Lenarz, Thomas; Lim, Hubert H

2015-01-01

Considerable progress has been made in the treatment of hearing loss with auditory implants. However, there are still many implanted patients that experience hearing deficiencies, such as limited speech understanding or vanishing perception with continuous stimulation (i.e., abnormal loudness adaptation). The present study aims to identify specific patterns of cerebral cortex activity involved with such deficiencies. We performed O-15-water positron emission tomography (PET) in patients implanted with electrodes within the cochlea, brainstem, or midbrain to investigate the pattern of cortical activation in response to speech or continuous multi-tone stimuli directly inputted into the implant processor that then delivered electrical patterns through those electrodes. Statistical parametric mapping was performed on a single subject basis. Better speech understanding was correlated with a larger extent of bilateral auditory cortex activation. In contrast to speech, the continuous multi-tone stimulus elicited mainly unilateral auditory cortical activity in which greater loudness adaptation corresponded to weaker activation and even deactivation. Interestingly, greater loudness adaptation was correlated with stronger activity within the ventral prefrontal cortex, which could be up-regulated to suppress the irrelevant or aberrant signals into the auditory cortex. The ability to detect these specific cortical patterns and differences across patients and stimuli demonstrates the potential for using PET to diagnose auditory function or dysfunction in implant patients, which in turn could guide the development of appropriate stimulation strategies for improving hearing rehabilitation. Beyond hearing restoration, our study also reveals a potential role of the frontal cortex in suppressing irrelevant or aberrant activity within the auditory cortex, and thus may be relevant for understanding and treating tinnitus.

Positron Emission Tomography Imaging Reveals Auditory and Frontal Cortical Regions Involved with Speech Perception and Loudness Adaptation

PubMed Central

Berding, Georg; Wilke, Florian; Rode, Thilo; Haense, Cathleen; Joseph, Gert; Meyer, Geerd J.; Mamach, Martin; Lenarz, Minoo; Geworski, Lilli; Bengel, Frank M.; Lenarz, Thomas; Lim, Hubert H.

2015-01-01

Considerable progress has been made in the treatment of hearing loss with auditory implants. However, there are still many implanted patients that experience hearing deficiencies, such as limited speech understanding or vanishing perception with continuous stimulation (i.e., abnormal loudness adaptation). The present study aims to identify specific patterns of cerebral cortex activity involved with such deficiencies. We performed O-15-water positron emission tomography (PET) in patients implanted with electrodes within the cochlea, brainstem, or midbrain to investigate the pattern of cortical activation in response to speech or continuous multi-tone stimuli directly inputted into the implant processor that then delivered electrical patterns through those electrodes. Statistical parametric mapping was performed on a single subject basis. Better speech understanding was correlated with a larger extent of bilateral auditory cortex activation. In contrast to speech, the continuous multi-tone stimulus elicited mainly unilateral auditory cortical activity in which greater loudness adaptation corresponded to weaker activation and even deactivation. Interestingly, greater loudness adaptation was correlated with stronger activity within the ventral prefrontal cortex, which could be up-regulated to suppress the irrelevant or aberrant signals into the auditory cortex. The ability to detect these specific cortical patterns and differences across patients and stimuli demonstrates the potential for using PET to diagnose auditory function or dysfunction in implant patients, which in turn could guide the development of appropriate stimulation strategies for improving hearing rehabilitation. Beyond hearing restoration, our study also reveals a potential role of the frontal cortex in suppressing irrelevant or aberrant activity within the auditory cortex, and thus may be relevant for understanding and treating tinnitus. PMID:26046763

Development and performance of power processor system for 2-gigahertz, 200-watt amplifier for communications technology satellite

NASA Technical Reports Server (NTRS)

Siegert, C. E.; Gourash, F.; Vasicek, R. W.

1977-01-01

The electrical and environmental requirements for a power processor system (PPS) designed to supply the appropriate voltages and currents to a 200-watt traveling wave tube (TWT) for a communication technology satellite is described. A block diagram of the PPS, the interface requirements between the PPS and spacecraft, the interface requirements between the PPS and 200-watt TWT, and the environmental requirements of the PPS are presented. Also included are discussions of protection circuits, interlocking sequences, and transient requirements. Predictions of the flight performance, based on ground test data, are provided.

JPRS Report, Science & Technology, USSR: Computers, Control Systems and Machines

DTIC Science & Technology

1989-03-14

optimizatsii slozhnykh sistem (Coding Theory and Complex System Optimization ). Alma-Ata, Nauka Press, 1977, pp. 8-16. 11. Author’s certificate number...Interpreter Specifics [0. I. Amvrosova] ............................................. 141 Creation of Modern Computer Systems for Complex Ecological...processor can be designed to decrease degradation upon failure and assure more reliable processor operation, without requiring more complex software or

Broadband set-top box using MAP-CA processor

NASA Astrophysics Data System (ADS)

Bush, John E.; Lee, Woobin; Basoglu, Chris

2001-12-01

Advances in broadband access are expected to exert a profound impact in our everyday life. It will be the key to the digital convergence of communication, computer and consumer equipment. A common thread that facilitates this convergence comprises digital media and Internet. To address this market, Equator Technologies, Inc., is developing the Dolphin broadband set-top box reference platform using its MAP-CA Broadband Signal ProcessorT chip. The Dolphin reference platform is a universal media platform for display and presentation of digital contents on end-user entertainment systems. The objective of the Dolphin reference platform is to provide a complete set-top box system based on the MAP-CA processor. It includes all the necessary hardware and software components for the emerging broadcast and the broadband digital media market based on IP protocols. Such reference design requires a broadband Internet access and high-performance digital signal processing. By using the MAP-CA processor, the Dolphin reference platform is completely programmable, allowing various codecs to be implemented in software, such as MPEG-2, MPEG-4, H.263 and proprietary codecs. The software implementation also enables field upgrades to keep pace with evolving technology and industry demands.

Speech, stone tool-making and the evolution of language.

PubMed

Cataldo, Dana Michelle; Migliano, Andrea Bamberg; Vinicius, Lucio

2018-01-01

The 'technological hypothesis' proposes that gestural language evolved in early hominins to enable the cultural transmission of stone tool-making skills, with speech appearing later in response to the complex lithic industries of more recent hominins. However, no flintknapping study has assessed the efficiency of speech alone (unassisted by gesture) as a tool-making transmission aid. Here we show that subjects instructed by speech alone underperform in stone tool-making experiments in comparison to subjects instructed through either gesture alone or 'full language' (gesture plus speech), and also report lower satisfaction with their received instruction. The results provide evidence that gesture was likely to be selected over speech as a teaching aid in the earliest hominin tool-makers; that speech could not have replaced gesturing as a tool-making teaching aid in later hominins, possibly explaining the functional retention of gesturing in the full language of modern humans; and that speech may have evolved for reasons unrelated to tool-making. We conclude that speech is unlikely to have evolved as tool-making teaching aid superior to gesture, as claimed by the technological hypothesis, and therefore alternative views should be considered. For example, gestural language may have evolved to enable tool-making in earlier hominins, while speech may have later emerged as a response to increased trade and more complex inter- and intra-group interactions in Middle Pleistocene ancestors of Neanderthals and Homo sapiens; or gesture and speech may have evolved in parallel rather than in sequence.

A Fully-Implantable Cochlear Implant SoC with Piezoelectric Middle-Ear Sensor and Arbitrary Waveform Neural Stimulation.

PubMed

Yip, Marcus; Jin, Rui; Nakajima, Hideko Heidi; Stankovic, Konstantina M; Chandrakasan, Anantha P

2015-01-01

A system-on-chip for an invisible, fully-implantable cochlear implant is presented. Implantable acoustic sensing is achieved by interfacing the SoC to a piezoelectric sensor that detects the sound-induced motion of the middle ear. Measurements from human cadaveric ears demonstrate that the sensor can detect sounds between 40 and 90 dB SPL over the speech bandwidth. A highly-reconfigurable digital sound processor enables system power scalability by reconfiguring the number of channels, and provides programmable features to enable a patient-specific fit. A mixed-signal arbitrary waveform neural stimulator enables energy-optimal stimulation pulses to be delivered to the auditory nerve. The energy-optimal waveform is validated with in-vivo measurements from four human subjects which show a 15% to 35% energy saving over the conventional rectangular waveform. Prototyped in a 0.18 μ m high-voltage CMOS technology, the SoC in 8-channel mode consumes 572 μ W of power including stimulation. The SoC integrates implantable acoustic sensing, sound processing, and neural stimulation on one chip to minimize the implant size, and proof-of-concept is demonstrated with measurements from a human cadaver ear.

Micropower Mixed-signal VLSI Independent Component Analysis for Gradient Flow Acoustic Source Separation.

PubMed

Stanaćević, Milutin; Li, Shuo; Cauwenberghs, Gert

2016-07-01

A parallel micro-power mixed-signal VLSI implementation of independent component analysis (ICA) with reconfigurable outer-product learning rules is presented. With the gradient sensing of the acoustic field over a miniature microphone array as a pre-processing method, the proposed ICA implementation can separate and localize up to 3 sources in mild reverberant environment. The ICA processor is implemented in 0.5 µm CMOS technology and occupies 3 mm × 3 mm area. At 16 kHz sampling rate, ASIC consumes 195 µW power from a 3 V supply. The outer-product implementation of natural gradient and Herault-Jutten ICA update rules demonstrates comparable performance to benchmark FastICA algorithm in ideal conditions and more robust performance in noisy and reverberant environment. Experiments demonstrate perceptually clear separation and precise localization over wide range of separation angles of two speech sources presented through speakers positioned at 1.5 m from the array on a conference room table. The presented ASIC leads to a extreme small form factor and low power consumption microsystem for source separation and localization required in applications like intelligent hearing aids and wireless distributed acoustic sensor arrays.

Auditory Environment Across the Life Span of Cochlear Implant Users: Insights From Data Logging.

PubMed

Busch, Tobias; Vanpoucke, Filiep; van Wieringen, Astrid

2017-05-24

We describe the natural auditory environment of people with cochlear implants (CIs), how it changes across the life span, and how it varies between individuals. We performed a retrospective cross-sectional analysis of Cochlear Nucleus 6 CI sound-processor data logs. The logs were obtained from 1,501 people with CIs (ages 0-96 years). They covered over 2.4 million hr of implant use and indicated how much time the CI users had spent in various acoustical environments. We investigated exposure to spoken language, noise, music, and quiet, and analyzed variation between age groups, users, and countries. CI users spent a substantial part of their daily life in noisy environments. As a consequence, most speech was presented in background noise. We found significant differences between age groups for all auditory scenes. Yet even within the same age group and country, variability between individuals was substantial. Regardless of their age, people with CIs face challenging acoustical environments in their daily life. Our results underline the importance of supporting them with assistive listening technology. Moreover, we found large differences between individuals' auditory diets that might contribute to differences in rehabilitation outcomes. Their causes and effects should be investigated further.

WATERLOPP V2/64: A highly parallel machine for numerical computation

NASA Astrophysics Data System (ADS)

Ostlund, Neil S.

1985-07-01

Current technological trends suggest that the high performance scientific machines of the future are very likely to consist of a large number (greater than 1024) of processors connected and communicating with each other in some as yet undetermined manner. Such an assembly of processors should behave as a single machine in obtaining numerical solutions to scientific problems. However, the appropriate way of organizing both the hardware and software of such an assembly of processors is an unsolved and active area of research. It is particularly important to minimize the organizational overhead of interprocessor comunication, global synchronization, and contention for shared resources if the performance of a large number ( n) of processors is to be anything like the desirable n times the performance of a single processor. In many situations, adding a processor actually decreases the performance of the overall system since the extra organizational overhead is larger than the extra processing power added. The systolic loop architecture is a new multiple processor architecture which attemps at a solution to the problem of how to organize a large number of asynchronous processors into an effective computational system while minimizing the organizational overhead. This paper gives a brief overview of the basic systolic loop architecture, systolic loop algorithms for numerical computation, and a 64-processor implementation of the architecture, WATERLOOP V2/64, that is being used as a testbed for exploring the hardware, software, and algorithmic aspects of the architecture.

The effect of hearing aid technologies on listening in an automobile.

PubMed

Wu, Yu-Hsiang; Stangl, Elizabeth; Bentler, Ruth A; Stanziola, Rachel W

2013-06-01

Communication while traveling in an automobile often is very difficult for hearing aid users. This is because the automobile/road noise level is usually high, and listeners/drivers often do not have access to visual cues. Since the talker of interest usually is not located in front of the listener/driver, conventional directional processing that places the directivity beam toward the listener's front may not be helpful and, in fact, could have a negative impact on speech recognition (when compared to omnidirectional processing). Recently, technologies have become available in commercial hearing aids that are designed to improve speech recognition and/or listening effort in noisy conditions where talkers are located behind or beside the listener. These technologies include (1) a directional microphone system that uses a backward-facing directivity pattern (Back-DIR processing), (2) a technology that transmits audio signals from the ear with the better signal-to-noise ratio (SNR) to the ear with the poorer SNR (Side-Transmission processing), and (3) a signal processing scheme that suppresses the noise at the ear with the poorer SNR (Side-Suppression processing). The purpose of the current study was to determine the effect of (1) conventional directional microphones and (2) newer signal processing schemes (Back-DIR, Side-Transmission, and Side-Suppression) on listener's speech recognition performance and preference for communication in a traveling automobile. A single-blinded, repeated-measures design was used. Twenty-five adults with bilateral symmetrical sensorineural hearing loss aged 44 through 84 yr participated in the study. The automobile/road noise and sentences of the Connected Speech Test (CST) were recorded through hearing aids in a standard van moving at a speed of 70 mph on a paved highway. The hearing aids were programmed to omnidirectional microphone, conventional adaptive directional microphone, and the three newer schemes. CST sentences were presented from the side and back of the hearing aids, which were placed on the ears of a manikin. The recorded stimuli were presented to listeners via earphones in a sound-treated booth to assess speech recognition performance and preference with each programmed condition. Compared to omnidirectional microphones, conventional adaptive directional processing had a detrimental effect on speech recognition when speech was presented from the back or side of the listener. Back-DIR and Side-Transmission processing improved speech recognition performance (relative to both omnidirectional and adaptive directional processing) when speech was from the back and side, respectively. The performance with Side-Suppression processing was better than with adaptive directional processing when speech was from the side. The participants' preferences for a given processing scheme were generally consistent with speech recognition results. The finding that performance with adaptive directional processing was poorer than with omnidirectional microphones demonstrates the importance of selecting the correct microphone technology for different listening situations. The results also suggest the feasibility of using hearing aid technologies to provide a better listening experience for hearing aid users in automobiles. American Academy of Audiology.

Creative Speech Technology: editorial introduction to this special issue.

PubMed

Edwards, Alistair D N; Newell, Christopher

2013-10-01

CreST is the Creative Speech Technology Network, a research network which brought together people from a wide variety of backgrounds spanning arts technology and beyond. The papers in this volume represent some of the outcomes of that collaboration. This editorial introduces the background of the network and each of the papers. In conclusion we demonstrate that this work helped to realize many of the objectives of the network.

Binaural hearing with electrical stimulation.

PubMed

Kan, Alan; Litovsky, Ruth Y

2015-04-01

Bilateral cochlear implantation is becoming a standard of care in many clinics. While much benefit has been shown through bilateral implantation, patients who have bilateral cochlear implants (CIs) still do not perform as well as normal hearing listeners in sound localization and understanding speech in noisy environments. This difference in performance can arise from a number of different factors, including the areas of hardware and engineering, surgical precision and pathology of the auditory system in deaf persons. While surgical precision and individual pathology are factors that are beyond careful control, improvements can be made in the areas of clinical practice and the engineering of binaural speech processors. These improvements should be grounded in a good understanding of the sensitivities of bilateral CI patients to the acoustic binaural cues that are important to normal hearing listeners for sound localization and speech in noise understanding. To this end, we review the current state-of-the-art in the understanding of the sensitivities of bilateral CI patients to binaural cues in electric hearing, and highlight the important issues and challenges as they relate to clinical practice and the development of new binaural processing strategies. This article is part of a Special Issue entitled . Copyright © 2014 Elsevier B.V. All rights reserved.

A clinical protocol for predicting outcomes with an implantable prosthetic device (Baha) in patients with single-sided deafness.

PubMed

Snapp, Hillary A; Fabry, David A; Telischi, Fred F; Arheart, Kristopher L; Angeli, Simon I

2010-01-01

The Baha implant is increasingly becoming a common form of treatment for individuals with single-sided deafness (SSD). However, evidence-based guidelines for determining candidacy in these patients are not yet established. The purpose of this study was to investigate the clinical utility of speech-in-noise testing as a part of the preoperative evaluation of the Baha device in patients with SSD. The study design was a prospective cohort of 24 English-speaking adults comparing preoperative results on speech-in-noise measures using the Baha Cordelle II headband stimulator to postoperative results using the patient's external Baha processor. Outcome measures included signal-to-noise ratio (SNR) loss as measured by the QuickSIN™ and scores of self-reported disability questionnaires. Wilcoxon signed-rank test resulted in no significant difference between the preoperative and postoperative methods for measuring benefit on listening in noise tasks. Passing Bablok regression analysis showed the preoperative and postoperative results to be statistically equivalent, which suggests that postoperative results can be predicted during preoperative testing. Wilcoxon signed-rank test showed significant improvements in self-reported disability postoperatively. The results support the use of speech-in-noise measures as an accurate predictor of overall benefit in patients with SSD prior to implantation. American Academy of Audiology.

Visualizing Syllables: Real-Time Computerized Feedback within a Speech-Language Intervention

ERIC Educational Resources Information Center

DeThorne, Laura; Aparicio Betancourt, Mariana; Karahalios, Karrie; Halle, Jim; Bogue, Ellen

2015-01-01

Computerized technologies now offer unprecedented opportunities to provide real-time visual feedback to facilitate children's speech-language development. We employed a mixed-method design to examine the effectiveness of two speech-language interventions aimed at facilitating children's multisyllabic productions: one incorporated a novel…

Automatic Speech Recognition Technology as an Effective Means for Teaching Pronunciation

ERIC Educational Resources Information Center

Elimat, Amal Khalil; AbuSeileek, Ali Farhan

2014-01-01

This study aimed to explore the effect of using automatic speech recognition technology (ASR) on the third grade EFL students' performance in pronunciation, whether teaching pronunciation through ASR is better than regular instruction, and the most effective teaching technique (individual work, pair work, or group work) in teaching pronunciation…

The Promise of NLP and Speech Processing Technologies in Language Assessment

ERIC Educational Resources Information Center

Chapelle, Carol A.; Chung, Yoo-Ree

2010-01-01

Advances in natural language processing (NLP) and automatic speech recognition and processing technologies offer new opportunities for language testing. Despite their potential uses on a range of language test item types, relatively little work has been done in this area, and it is therefore not well understood by test developers, researchers or…

Multiple Stakeholder Perspectives on Teletherapy Delivery of Speech Pathology Services in Rural Schools: A Preliminary, Qualitative Investigation

PubMed Central

LINCOLN, MICHELLE; HINES, MONIQUE; FAIRWEATHER, CRAIG; RAMSDEN, ROBYN; MARTINOVICH, JULIA

2015-01-01

The objective of this study was to investigate stakeholders’ views on the feasibility and acceptability of a pilot speech pathology teletherapy program for children attending schools in rural New South Wales, Australia. Nine children received speech pathology sessions delivered via Adobe Connect® web-conferencing software. During semi-structured interviews, school principals (n = 3), therapy facilitators (n = 7), and parents (n = 6) described factors that promoted or threatened the program’s feasibility and acceptability. Themes were categorized according to whether they related to (a) the use of technology; (b) the school-based nature of the program; or (c) the combination of using technology with a school-based program. Despite frequent reports of difficulties with technology, teletherapy delivery of speech pathology services in schools was highly acceptable to stakeholders. However, the use of technology within a school environment increased the complexities of service delivery. Service providers should pay careful attention to planning processes and lines of communication in order to promote efficiency and acceptability of teletherapy programs. PMID:25945230

Computational needs survey of NASA automation and robotics missions. Volume 1: Survey and results

NASA Technical Reports Server (NTRS)

Davis, Gloria J.

1991-01-01

NASA's operational use of advanced processor technology in space systems lags behind its commercial development by more than eight years. One of the factors contributing to this is that mission computing requirements are frequently unknown, unstated, misrepresented, or simply not available in a timely manner. NASA must provide clear common requirements to make better use of available technology, to cut development lead time on deployable architectures, and to increase the utilization of new technology. A preliminary set of advanced mission computational processing requirements of automation and robotics (A&R) systems are provided for use by NASA, industry, and academic communities. These results were obtained in an assessment of the computational needs of current projects throughout NASA. The high percent of responses indicated a general need for enhanced computational capabilities beyond the currently available 80386 and 68020 processor technology. Because of the need for faster processors and more memory, 90 percent of the polled automation projects have reduced or will reduce the scope of their implementation capabilities. The requirements are presented with respect to their targeted environment, identifying the applications required, system performance levels necessary to support them, and the degree to which they are met with typical programmatic constraints. Volume one includes the survey and results. Volume two contains the appendixes.

Computational needs survey of NASA automation and robotics missions. Volume 2: Appendixes

NASA Technical Reports Server (NTRS)

Davis, Gloria J.

1991-01-01

NASA's operational use of advanced processor technology in space systems lags behind its commercial development by more than eight years. One of the factors contributing to this is the fact that mission computing requirements are frequency unknown, unstated, misrepresented, or simply not available in a timely manner. NASA must provide clear common requirements to make better use of available technology, to cut development lead time on deployable architectures, and to increase the utilization of new technology. Here, NASA, industry and academic communities are provided with a preliminary set of advanced mission computational processing requirements of automation and robotics (A and R) systems. The results were obtained in an assessment of the computational needs of current projects throughout NASA. The high percent of responses indicated a general need for enhanced computational capabilities beyond the currently available 80386 and 68020 processor technology. Because of the need for faster processors and more memory, 90 percent of the polled automation projects have reduced or will reduce the scope of their implemented capabilities. The requirements are presented with respect to their targeted environment, identifying the applications required, system performance levels necessary to support them, and the degree to which they are met with typical programmatic constraints. Here, appendixes are provided.

Technology transfer of military space microprocessor developments

NASA Astrophysics Data System (ADS)

Gorden, C.; King, D.; Byington, L.; Lanza, D.

1999-01-01

Over the past 13 years the Air Force Research Laboratory (AFRL) has led the development of microprocessors and computers for USAF space and strategic missile applications. As a result of these Air Force development programs, advanced computer technology is available for use by civil and commercial space customers as well. The Generic VHSIC Spaceborne Computer (GVSC) program began in 1985 at AFRL to fulfill a deficiency in the availability of space-qualified data and control processors. GVSC developed a radiation hardened multi-chip version of the 16-bit, Mil-Std 1750A microprocessor. The follow-on to GVSC, the Advanced Spaceborne Computer Module (ASCM) program, was initiated by AFRL to establish two industrial sources for complete, radiation-hardened 16-bit and 32-bit computers and microelectronic components. Development of the Control Processor Module (CPM), the first of two ASCM contract phases, concluded in 1994 with the availability of two sources for space-qualified, 16-bit Mil-Std-1750A computers, cards, multi-chip modules, and integrated circuits. The second phase of the program, the Advanced Technology Insertion Module (ATIM), was completed in December 1997. ATIM developed two single board computers based on 32-bit reduced instruction set computer (RISC) processors. GVSC, CPM, and ATIM technologies are flying or baselined into the majority of today's DoD, NASA, and commercial satellite systems.

Evaluation of Adaptive Noise Management Technologies for School-Age Children with Hearing Loss.

PubMed

Wolfe, Jace; Duke, Mila; Schafer, Erin; Jones, Christine; Rakita, Lori

2017-05-01

Children with hearing loss experience significant difficulty understanding speech in noisy and reverberant situations. Adaptive noise management technologies, such as fully adaptive directional microphones and digital noise reduction, have the potential to improve communication in noise for children with hearing aids. However, there are no published studies evaluating the potential benefits children receive from the use of adaptive noise management technologies in simulated real-world environments as well as in daily situations. The objective of this study was to compare speech recognition, speech intelligibility ratings (SIRs), and sound preferences of children using hearing aids equipped with and without adaptive noise management technologies. A single-group, repeated measures design was used to evaluate performance differences obtained in four simulated environments. In each simulated environment, participants were tested in a basic listening program with minimal noise management features, a manual program designed for that scene, and the hearing instruments' adaptive operating system that steered hearing instrument parameterization based on the characteristics of the environment. Twelve children with mild to moderately severe sensorineural hearing loss. Speech recognition and SIRs were evaluated in three hearing aid programs with and without noise management technologies across two different test sessions and various listening environments. Also, the participants' perceptual hearing performance in daily real-world listening situations with two of the hearing aid programs was evaluated during a four- to six-week field trial that took place between the two laboratory sessions. On average, the use of adaptive noise management technology improved sentence recognition in noise for speech presented in front of the participant but resulted in a decrement in performance for signals arriving from behind when the participant was facing forward. However, the improvement with adaptive noise management exceeded the decrement obtained when the signal arrived from behind. Most participants reported better subjective SIRs when using adaptive noise management technologies, particularly when the signal of interest arrived from in front of the listener. In addition, most participants reported a preference for the technology with an automatically switching, adaptive directional microphone and adaptive noise reduction in real-world listening situations when compared to conventional, omnidirectional microphone use with minimal noise reduction processing. Use of the adaptive noise management technologies evaluated in this study improves school-age children's speech recognition in noise for signals arriving from the front. Although a small decrement in speech recognition in noise was observed for signals arriving from behind the listener, most participants reported a preference for use of noise management technology both when the signal arrived from in front and from behind the child. The results of this study suggest that adaptive noise management technologies should be considered for use with school-age children when listening in academic and social situations. American Academy of Audiology

Dynamic behavior of gasoline fuel cell electric vehicles

NASA Astrophysics Data System (ADS)

Mitchell, William; Bowers, Brian J.; Garnier, Christophe; Boudjemaa, Fabien

As we begin the 21st century, society is continuing efforts towards finding clean power sources and alternative forms of energy. In the automotive sector, reduction of pollutants and greenhouse gas emissions from the power plant is one of the main objectives of car manufacturers and innovative technologies are under active consideration to achieve this goal. One technology that has been proposed and vigorously pursued in the past decade is the proton exchange membrane (PEM) fuel cell, an electrochemical device that reacts hydrogen with oxygen to produce water, electricity and heat. Since today there is no existing extensive hydrogen infrastructure and no commercially viable hydrogen storage technology for vehicles, there is a continuing debate as to how the hydrogen for these advanced vehicles will be supplied. In order to circumvent the above issues, power systems based on PEM fuel cells can employ an on-board fuel processor that has the ability to convert conventional fuels such as gasoline into hydrogen for the fuel cell. This option could thereby remove the fuel infrastructure and storage issues. However, for these fuel processor/fuel cell vehicles to be commercially successful, issues such as start time and transient response must be addressed. This paper discusses the role of transient response of the fuel processor power plant and how it relates to the battery sizing for a gasoline fuel cell vehicle. In addition, results of fuel processor testing from a current Renault/Nuvera Fuel Cells project are presented to show the progress in transient performance.

Programmable optical processor chips: toward photonic RF filters with DSP-level flexibility and MHz-band selectivity

NASA Astrophysics Data System (ADS)

Xie, Yiwei; Geng, Zihan; Zhuang, Leimeng; Burla, Maurizio; Taddei, Caterina; Hoekman, Marcel; Leinse, Arne; Roeloffzen, Chris G. H.; Boller, Klaus-J.; Lowery, Arthur J.

2017-12-01

Integrated optical signal processors have been identified as a powerful engine for optical processing of microwave signals. They enable wideband and stable signal processing operations on miniaturized chips with ultimate control precision. As a promising application, such processors enables photonic implementations of reconfigurable radio frequency (RF) filters with wide design flexibility, large bandwidth, and high-frequency selectivity. This is a key technology for photonic-assisted RF front ends that opens a path to overcoming the bandwidth limitation of current digital electronics. Here, the recent progress of integrated optical signal processors for implementing such RF filters is reviewed. We highlight the use of a low-loss, high-index-contrast stoichiometric silicon nitride waveguide which promises to serve as a practical material platform for realizing high-performance optical signal processors and points toward photonic RF filters with digital signal processing (DSP)-level flexibility, hundreds-GHz bandwidth, MHz-band frequency selectivity, and full system integration on a chip scale.

Design and development of an AAC app based on a speech-to-symbol technology.

PubMed

Radici, Elena; Bonacina, Stefano; De Leo, Gianluca

2016-08-01

The purpose of this paper is to present the design and the development of an Augmentative and Alternative Communication app that uses a speech to symbol technology to model language, i.e. to recognize the speech and display the text or the graphic content related to it. Our app is intended to be adopted by communication partners who want to engage in interventions focused on improving communication skills. Our app has the goal of translating simple speech sentences in a set of symbols that are understandable by children with complex communication needs. We moderated a focus group among six AAC communication partners. Then, we developed a prototype. We are currently starting testing our app in an AAC Centre in Milan, Italy.

Choosing and Using Text-to-Speech Software

ERIC Educational Resources Information Center

Peters, Tom; Bell, Lori

2007-01-01

This article describes a computer-based technology for generating speech called text-to-speech (TTS). This software is ready for widespread use by libraries, other organizations, and individual users. It offers the affordable ability to turn just about any electronic text that is not image-based into an artificially spoken communication. The…

Digital Data Collection and Analysis: Application for Clinical Practice

ERIC Educational Resources Information Center

Ingram, Kelly; Bunta, Ferenc; Ingram, David

2004-01-01

Technology for digital speech recording and speech analysis is now readily available for all clinicians who use a computer. This article discusses some advantages of moving from analog to digital recordings and outlines basic recording procedures. The purpose of this article is to familiarize speech-language pathologists with computerized audio…

A comparison of aphasic and non-brain-injured adults on a dichotic CV-syllable listening task.

PubMed

Shanks, J; Ryan, W

1976-06-01

A dichotic CV-syllable listening task was administered to a group of eleven non-brain-injured adults and to a group of eleven adult aphasics. The results of this study may be summarized as follows: 1)The group of non-brain-injured adults showed a slight right ear advantage for dichotically presented CV-syllables. 2)In comparison with the control group the asphasic group showed a bilateral deficit in response to the dichotic CV-syllables, superimposed on a non-significant right ear advantage. 3) The asphasic group demonstrated a great deal of intersubject variability on the dichotic task with six aphasics showing a right ear preference for the stimuli. The non-brain-injured subjects performed more homogeneously on the task. 4) The two subgroups of aphasics, a right ear advantage group and a left ear advantage group, performed significantly different on the dichotic listening task. 5) Single correct data analysis proved valuable by deleting accuracy of report for an examination of trials in which there was true competition for the single left hemispheric speech processor. These results were analyzed in terms of a functional model of auditory processing. In view of this model, the bilateral deficit in dichotic performance of the asphasic group was accounted for by the presence of a lesion within the dominant left hemisphere, where the speech signals from both ears converge for final processing. The right ear advantage shown by one asphasic subgroup was explained by a lesion interfering with the corpus callosal pathways from the left hemisphere; the left ear advantage observed within the other subgroup was explained by a lesion in the area of the auditory processor of the left hemisphere.

Assessment of SAR and thermal changes near a cochlear implant system for mobile phone type exposures.

PubMed

McIntosh, Robert L; Iskra, Steve; McKenzie, Raymond J; Chambers, John; Metzenthen, Bill; Anderson, Vitas

2008-01-01

A cochlear implant system is a device used to enable hearing in people with severe hearing loss and consists of an internal implant and external speech processor. This study considers the effect of scattered radiofrequency fields when these persons are subject to mobile phone type exposure. A worst-case scenario is considered where the antenna is operating at nominal full power, the speech processor is situated behind the ear using a metallic hook, and the antenna is adjacent to the hook and the internal ball electrode. The resultant energy deposition and thermal changes were determined through numerical modelling. With a 900 MHz half-wave dipole antenna producing continuous-wave (CW) 250 mW power, the maximum 10 g averaged SAR was 1.31 W/kg which occurred in the vicinity of the hook and the ball electrode. The maximum temperature increase was 0.33 degrees C in skin adjacent to the hook. For the 1800 MHz antenna, operating at 125 mW, the maximum 10 g averaged SAR was 0.93 W/kg in the pinna whilst the maximum temperature change was 0.16 degrees C. The analysis predicts that the wearer complies with the radiofrequency safety limits specified by the International Commission on Non-Ionizing Radiation Protection (ICNIRP), the Institute of Electrical and Electronics Engineers (IEEE), and the Australian Radiation Protection and Nuclear Safety Agency (ARPANSA) for 900 and 1800 MHz mobile phone type exposure and thus raises no cause for concern. The resultant temperature increase is well below the maximum rise of 1 degrees C recommended by ICNIRP. Effects in the cochlea were insignificant. (c) 2007 Wiley-Liss, Inc.

New Dimensions in Microarchitecture Harnessing 3D Integration Technologies (BRIEFING CHARTS)

DTIC Science & Technology

2007-03-06

Quad Core Bandwidth and Latency Boundaries General Purpose Processor Loads Latency limited Ba nd w id th li m ite dProcessor load trade -off between I...delay No= number of ckts at 1V do= ckt delay at 1V From “3D Intergration ” Special Topic Sessionl W. Haensch, ISSCC ‘07, 2/07 11 DARPA MTS March 6, 2007

Fuel Processor Development for a Soldier-Portable Fuel Cell System

DOE Office of Scientific and Technical Information (OSTI.GOV)

Palo, Daniel R.; Holladay, Jamie D.; Rozmiarek, Robert T.

2002-01-01

Battelle is currently developing a soldier-portable power system for the U.S. Army that will continuously provide 15 W (25 W peak) of base load electric power for weeks or months using a micro technology-based fuel processor. The fuel processing train consists of a combustor, two vaporizers, and a steam-reforming reactor. This paper describes the concept and experimental progress to date.

Speech Recognition as a Support Service for Deaf and Hard of Hearing Students: Adaptation and Evaluation. Final Report to Spencer Foundation.

ERIC Educational Resources Information Center

Stinson, Michael; Elliot, Lisa; McKee, Barbara; Coyne, Gina

This report discusses a project that adapted new automatic speech recognition (ASR) technology to provide real-time speech-to-text transcription as a support service for students who are deaf and hard of hearing (D/HH). In this system, as the teacher speaks, a hearing intermediary, or captionist, dictates into the speech recognition system in a…

Is talking to an automated teller machine natural and fun?

PubMed

Chan, F Y; Khalid, H M

Usability and affective issues of using automatic speech recognition technology to interact with an automated teller machine (ATM) are investigated in two experiments. The first uncovered dialogue patterns of ATM users for the purpose of designing the user interface for a simulated speech ATM system. Applying the Wizard-of-Oz methodology, multiple mapping and word spotting techniques, the speech driven ATM accommodates bilingual users of Bahasa Melayu and English. The second experiment evaluates the usability of a hybrid speech ATM, comparing it with a simulated manual ATM. The aim is to investigate how natural and fun can talking to a speech ATM be for these first-time users. Subjects performed the withdrawal and balance enquiry tasks. The ANOVA was performed on the usability and affective data. The results showed significant differences between systems in the ability to complete the tasks as well as in transaction errors. Performance was measured on the time taken by subjects to complete the task and the number of speech recognition errors that occurred. On the basis of user emotions, it can be said that the hybrid speech system enabled pleasurable interaction. Despite the limitations of speech recognition technology, users are set to talk to the ATM when it becomes available for public use.

Riding the Technology Wave.

ERIC Educational Resources Information Center

Malan, Pierre

This paper presents an overview of information technology development. The first section sets the scene, comparing the first WAN (Wide Area Network) and Intel processor to current technology. The birth of the microcomputer is described in the second section, including historical background on semiconductors, microprocessors, and the microcomputer.…

An exploratory study on the driving method of speech synthesis based on the human eye reading imaging data

NASA Astrophysics Data System (ADS)

Gao, Pei-pei; Liu, Feng

2016-10-01

With the development of information technology and artificial intelligence, speech synthesis plays a significant role in the fields of Human-Computer Interaction Techniques. However, the main problem of current speech synthesis techniques is lacking of naturalness and expressiveness so that it is not yet close to the standard of natural language. Another problem is that the human-computer interaction based on the speech synthesis is too monotonous to realize mechanism of user subjective drive. This thesis introduces the historical development of speech synthesis and summarizes the general process of this technique. It is pointed out that prosody generation module is an important part in the process of speech synthesis. On the basis of further research, using eye activity rules when reading to control and drive prosody generation was introduced as a new human-computer interaction method to enrich the synthetic form. In this article, the present situation of speech synthesis technology is reviewed in detail. Based on the premise of eye gaze data extraction, using eye movement signal in real-time driving, a speech synthesis method which can express the real speech rhythm of the speaker is proposed. That is, when reader is watching corpora with its eyes in silent reading, capture the reading information such as the eye gaze duration per prosodic unit, and establish a hierarchical prosodic pattern of duration model to determine the duration parameters of synthesized speech. At last, after the analysis, the feasibility of the above method is verified.

Implementation of the Intelligent Voice System for Kazakh

NASA Astrophysics Data System (ADS)

Yessenbayev, Zh; Saparkhojayev, N.; Tibeyev, T.

2014-04-01

Modern speech technologies are highly advanced and widely used in day-to-day applications. However, this is mostly concerned with the languages of well-developed countries such as English, German, Japan, Russian, etc. As for Kazakh, the situation is less prominent and research in this field is only starting to evolve. In this research and application-oriented project, we introduce an intelligent voice system for the fast deployment of call-centers and information desks supporting Kazakh speech. The demand on such a system is obvious if the country's large size and small population is considered. The landline and cell phones become the only means of communication for the distant villages and suburbs. The system features Kazakh speech recognition and synthesis modules as well as a web-GUI for efficient dialog management. For speech recognition we use CMU Sphinx engine and for speech synthesis- MaryTTS. The web-GUI is implemented in Java enabling operators to quickly create and manage the dialogs in user-friendly graphical environment. The call routines are handled by Asterisk PBX and JBoss Application Server. The system supports such technologies and protocols as VoIP, VoiceXML, FastAGI, Java SpeechAPI and J2EE. For the speech recognition experiments we compiled and used the first Kazakh speech corpus with the utterances from 169 native speakers. The performance of the speech recognizer is 4.1% WER on isolated word recognition and 6.9% WER on clean continuous speech recognition tasks. The speech synthesis experiments include the training of male and female voices.

Library Automation Design for Visually Impaired People

ERIC Educational Resources Information Center

Yurtay, Nilufer; Bicil, Yucel; Celebi, Sait; Cit, Guluzar; Dural, Deniz

2011-01-01

Speech synthesis is a technology used in many different areas in computer science. This technology can bring a solution to reading activity of visually impaired people due to its text to speech conversion. Based on this problem, in this study, a system is designed needed for a visually impaired person to make use of all the library facilities in…

Impact of Technology Based Instruction on Speech Competency and Presentation Confidence Levels of Hispanic College Students

ERIC Educational Resources Information Center

Mundy, Marie-Anne; Padilla Oviedo, Andres; Ramirez, Juan; Taylor, Nick; Flores, Itza

2014-01-01

One of the main goals of universities is to graduate students who are capable and competent in competing in the workforce. As presentational communication skills are critical in today's job market, Hispanic university students need to be trained to effectively develop and deliver presentational speeches. Web/technology enhanced training techniques…

Transcription and Annotation of a Japanese Accented Spoken Corpus of L2 Spanish for the Development of CAPT Applications

ERIC Educational Resources Information Center

Carranza, Mario

2016-01-01

This paper addresses the process of transcribing and annotating spontaneous non-native speech with the aim of compiling a training corpus for the development of Computer Assisted Pronunciation Training (CAPT) applications, enhanced with Automatic Speech Recognition (ASR) technology. To better adapt ASR technology to CAPT tools, the recognition…

FPGA wavelet processor design using language for instruction-set architectures (LISA)

NASA Astrophysics Data System (ADS)

Meyer-Bäse, Uwe; Vera, Alonzo; Rao, Suhasini; Lenk, Karl; Pattichis, Marios

2007-04-01

The design of an microprocessor is a long, tedious, and error-prone task consisting of typically three design phases: architecture exploration, software design (assembler, linker, loader, profiler), architecture implementation (RTL generation for FPGA or cell-based ASIC) and verification. The Language for instruction-set architectures (LISA) allows to model a microprocessor not only from instruction-set but also from architecture description including pipelining behavior that allows a design and development tool consistency over all levels of the design. To explore the capability of the LISA processor design platform a.k.a. CoWare Processor Designer we present in this paper three microprocessor designs that implement a 8/8 wavelet transform processor that is typically used in today's FBI fingerprint compression scheme. We have designed a 3 stage pipelined 16 bit RISC processor (NanoBlaze). Although RISC μPs are usually considered "fast" processors due to design concept like constant instruction word size, deep pipelines and many general purpose registers, it turns out that DSP operations consume essential processing time in a RISC processor. In a second step we have used design principles from programmable digital signal processor (PDSP) to improve the throughput of the DWT processor. A multiply-accumulate operation along with indirect addressing operation were the key to achieve higher throughput. A further improvement is possible with today's FPGA technology. Today's FPGAs offer a large number of embedded array multipliers and it is now feasible to design a "true" vector processor (TVP). A multiplication of two vectors can be done in just one clock cycle with our TVP, a complete scalar product in two clock cycles. Code profiling and Xilinx FPGA ISE synthesis results are provided that demonstrate the essential improvement that a TVP has compared with traditional RISC or PDSP designs.

Technology and the evolution of clinical methods for stuttering.

PubMed

Packman, Ann; Meredith, Grant

2011-06-01

The World Wide Web (WWW) was 20 years old last year. Enormous amounts of information about stuttering are now available to anyone who can access the Internet. Compared to 20 years ago, people who stutter and their families can now make more informed choices about speech-language interventions, from a distance. Blogs and chat rooms provide opportunities for people who stutter to share their experiences from a distance and to support one another. New technologies are also being adopted into speech-language pathology practice and service delivery. Telehealth is an exciting development as it means that treatment can now be made available to many rural and remotely located people who previously did not have access to it. Possible future technological developments for speech-language pathology practice include Internet based treatments and the use of Virtual Reality. Having speech and CBT treatments for stuttering available on the Internet would greatly increase their accessibility. Second Life also has exciting possibilities for people who stutter. The reader will (1) explain how people who stutter and their families can get information about stuttering from the World Wide Web, (2) discuss how new technologies have been applied in speech-language pathology practice, and (3) summarize the principles and practice of telehealth delivery of services for people who stutter and their families. Copyright © 2011. Published by Elsevier Inc.

Reprint of: technology and the evolution of clinical methods for stuttering.

PubMed

Packman, Ann; Meredith, Grant

2011-09-01

The World Wide Web (WWW) was 20 years old last year. Enormous amounts of information about stuttering are now available to anyone who can access the Internet. Compared to 20 years ago, people who stutter and their families can now make more informed choices about speech-language interventions, from a distance. Blogs and chat rooms provide opportunities for people who stutter to share their experiences from a distance and to support one another. New technologies are also being adopted into speech-language pathology practice and service delivery. Telehealth is an exciting development as it means that treatment can now be made available to many rural and remotely located people who previously did not have access to it. Possible future technological developments for speech-language pathology practice include Internet based treatments and the use of Virtual Reality. Having speech and CBT treatments for stuttering available on the Internet would greatly increase their accessibility. Second Life also has exciting possibilities for people who stutter. The reader will (1) explain how people who stutter and their families can get information about stuttering from the World Wide Web, (2) discuss how new technologies have been applied in speech-language pathology practice, and (3) summarize the principles and practice of telehealth delivery of services for people who stutter and their families. Copyright © 2011. Published by Elsevier Inc.

A Comparison of LBG and ADPCM Speech Compression Techniques

NASA Astrophysics Data System (ADS)

Bachu, Rajesh G.; Patel, Jignasa; Barkana, Buket D.

Speech compression is the technology of converting human speech into an efficiently encoded representation that can later be decoded to produce a close approximation of the original signal. In all speech there is a degree of predictability and speech coding techniques exploit this to reduce bit rates yet still maintain a suitable level of quality. This paper is a study and implementation of Linde-Buzo-Gray Algorithm (LBG) and Adaptive Differential Pulse Code Modulation (ADPCM) algorithms to compress speech signals. In here we implemented the methods using MATLAB 7.0. The methods we used in this study gave good results and performance in compressing the speech and listening tests showed that efficient and high quality coding is achieved.

A pipeline VLSI design of fast singular value decomposition processor for real-time EEG system based on on-line recursive independent component analysis.

PubMed

Huang, Kuan-Ju; Shih, Wei-Yeh; Chang, Jui Chung; Feng, Chih Wei; Fang, Wai-Chi

2013-01-01

This paper presents a pipeline VLSI design of fast singular value decomposition (SVD) processor for real-time electroencephalography (EEG) system based on on-line recursive independent component analysis (ORICA). Since SVD is used frequently in computations of the real-time EEG system, a low-latency and high-accuracy SVD processor is essential. During the EEG system process, the proposed SVD processor aims to solve the diagonal, inverse and inverse square root matrices of the target matrices in real time. Generally, SVD requires a huge amount of computation in hardware implementation. Therefore, this work proposes a novel design concept for data flow updating to assist the pipeline VLSI implementation. The SVD processor can greatly improve the feasibility of real-time EEG system applications such as brain computer interfaces (BCIs). The proposed architecture is implemented using TSMC 90 nm CMOS technology. The sample rate of EEG raw data adopts 128 Hz. The core size of the SVD processor is 580×580 um(2), and the speed of operation frequency is 20MHz. It consumes 0.774mW of power during the 8-channel EEG system per execution time.

Data preprocessing for determining outer/inner parallelization in the nested loop problem using OpenMP

NASA Astrophysics Data System (ADS)

Handhika, T.; Bustamam, A.; Ernastuti, Kerami, D.

2017-07-01

Multi-thread programming using OpenMP on the shared-memory architecture with hyperthreading technology allows the resource to be accessed by multiple processors simultaneously. Each processor can execute more than one thread for a certain period of time. However, its speedup depends on the ability of the processor to execute threads in limited quantities, especially the sequential algorithm which contains a nested loop. The number of the outer loop iterations is greater than the maximum number of threads that can be executed by a processor. The thread distribution technique that had been found previously only be applied by the high-level programmer. This paper generates a parallelization procedure for low-level programmer in dealing with 2-level nested loop problems with the maximum number of threads that can be executed by a processor is smaller than the number of the outer loop iterations. Data preprocessing which is related to the number of the outer loop and the inner loop iterations, the computational time required to execute each iteration and the maximum number of threads that can be executed by a processor are used as a strategy to determine which parallel region that will produce optimal speedup.

Micro-Based Speech Recognition: Instructional Innovation for Handicapped Learners.

ERIC Educational Resources Information Center

Horn, Carin E.; Scott, Brian L.

A new voice based learning system (VBLS), which allows the handicapped user to interact with a microcomputer by voice commands, is described. Speech or voice recognition is the computerized process of identifying a spoken word or phrase, including those resulting from speech impediments. This new technology is helpful to the severely physically…

Supporting Dictation Speech Recognition Error Correction: The Impact of External Information

ERIC Educational Resources Information Center

Shi, Yongmei; Zhou, Lina

2011-01-01

Although speech recognition technology has made remarkable progress, its wide adoption is still restricted by notable effort made and frustration experienced by users while correcting speech recognition errors. One of the promising ways to improve error correction is by providing user support. Although support mechanisms have been proposed for…

Using Web Speech Technology with Language Learning Applications

ERIC Educational Resources Information Center

Daniels, Paul

2015-01-01

In this article, the author presents the history of human-to-computer interaction based upon the design of sophisticated computerized speech recognition algorithms. Advancements such as the arrival of cloud-based computing and software like Google's Web Speech API allows anyone with an Internet connection and Chrome browser to take advantage of…

Underwater Threat Source Localization: Processing Sensor Network TDOAs with a Terascale Optical Core Device

DOE Office of Scientific and Technical Information (OSTI.GOV)

Barhen, Jacob; Imam, Neena

2007-01-01

Revolutionary computing technologies are defined in terms of technological breakthroughs, which leapfrog over near-term projected advances in conventional hardware and software to produce paradigm shifts in computational science. For underwater threat source localization using information provided by a dynamical sensor network, one of the most promising computational advances builds upon the emergence of digital optical-core devices. In this article, we present initial results of sensor network calculations that focus on the concept of signal wavefront time-difference-of-arrival (TDOA). The corresponding algorithms are implemented on the EnLight processing platform recently introduced by Lenslet Laboratories. This tera-scale digital optical core processor is optimizedmore » for array operations, which it performs in a fixed-point-arithmetic architecture. Our results (i) illustrate the ability to reach the required accuracy in the TDOA computation, and (ii) demonstrate that a considerable speed-up can be achieved when using the EnLight 64a prototype processor as compared to a dual Intel XeonTM processor.« less

The effect of hearing aid technologies on listening in an automobile

PubMed Central

Wu, Yu-Hsiang; Stangl, Elizabeth; Bentler, Ruth A.; Stanziola, Rachel W.

2014-01-01

Background Communication while traveling in an automobile often is very difficult for hearing aid users. This is because the automobile /road noise level is usually high, and listeners/drivers often do not have access to visual cues. Since the talker of interest usually is not located in front of the driver/listener, conventional directional processing that places the directivity beam toward the listener’s front may not be helpful, and in fact, could have a negative impact on speech recognition (when compared to omnidirectional processing). Recently, technologies have become available in commercial hearing aids that are designed to improve speech recognition and/or listening effort in noisy conditions where talkers are located behind or beside the listener. These technologies include (1) a directional microphone system that uses a backward-facing directivity pattern (Back-DIR processing), (2) a technology that transmits audio signals from the ear with the better signal-to-noise ratio (SNR) to the ear with the poorer SNR (Side-Transmission processing), and (3) a signal processing scheme that suppresses the noise at the ear with the poorer SNR (Side-Suppression processing). Purpose The purpose of the current study was to determine the effect of (1) conventional directional microphones and (2) newer signal processing schemes (Back-DIR, Side-Transmission, and Side-Suppression) on listener’s speech recognition performance and preference for communication in a traveling automobile. Research design A single-blinded, repeated-measures design was used. Study Sample Twenty-five adults with bilateral symmetrical sensorineural hearing loss aged 44 through 84 years participated in the study. Data Collection and Analysis The automobile/road noise and sentences of the Connected Speech Test (CST) were recorded through hearing aids in a standard van moving at a speed of 70 miles/hour on a paved highway. The hearing aids were programmed to omnidirectional microphone, conventional adaptive directional microphone, and the three newer schemes. CST sentences were presented from the side and back of the hearing aids, which were placed on the ears of a manikin. The recorded stimuli were presented to listeners via earphones in a sound treated booth to assess speech recognition performance and preference with each programmed condition. Results Compared to omnidirectional microphones, conventional adaptive directional processing had a detrimental effect on speech recognition when speech was presented from the back or side of the listener. Back-DIR and Side-Transmission processing improved speech recognition performance (relative to both omnidirectional and adaptive directional processing) when speech was from the back and side, respectively. The performance with Side-Suppression processing was better than with adaptive directional processing when speech was from the side. The participants’ preferences for a given processing scheme were generally consistent with speech recognition results. Conclusions The finding that performance with adaptive directional processing was poorer than with omnidirectional microphones demonstrates the importance of selecting the correct microphone technology for different listening situations. The results also suggest the feasibility of using hearing aid technologies to provide a better listening experience for hearing aid users in automobiles. PMID:23886425

The Speech multi features fusion perceptual hash algorithm based on tensor decomposition

NASA Astrophysics Data System (ADS)

Huang, Y. B.; Fan, M. H.; Zhang, Q. Y.

2018-03-01

With constant progress in modern speech communication technologies, the speech data is prone to be attacked by the noise or maliciously tampered. In order to make the speech perception hash algorithm has strong robustness and high efficiency, this paper put forward a speech perception hash algorithm based on the tensor decomposition and multi features is proposed. This algorithm analyses the speech perception feature acquires each speech component wavelet packet decomposition. LPCC, LSP and ISP feature of each speech component are extracted to constitute the speech feature tensor. Speech authentication is done by generating the hash values through feature matrix quantification which use mid-value. Experimental results showing that the proposed algorithm is robust for content to maintain operations compared with similar algorithms. It is able to resist the attack of the common background noise. Also, the algorithm is highly efficiency in terms of arithmetic, and is able to meet the real-time requirements of speech communication and complete the speech authentication quickly.

17 Ways to Say Yes: Toward Nuanced Tone of Voice in AAC and Speech Technology

PubMed Central

Pullin, Graham; Hennig, Shannon

2015-01-01

Abstract People with complex communication needs who use speech-generating devices have very little expressive control over their tone of voice. Despite its importance in human interaction, the issue of tone of voice remains all but absent from AAC research and development however. In this paper, we describe three interdisciplinary projects, past, present and future: The critical design collection Six Speaking Chairs has provoked deeper discussion and inspired a social model of tone of voice; the speculative concept Speech Hedge illustrates challenges and opportunities in designing more expressive user interfaces; the pilot project Tonetable could enable participatory research and seed a research network around tone of voice. We speculate that more radical interactions might expand frontiers of AAC and disrupt speech technology as a whole. PMID:25965913

Synthetic Aperture Radar (SAR) data processing

NASA Technical Reports Server (NTRS)

Beckner, F. L.; Ahr, H. A.; Ausherman, D. A.; Cutrona, L. J.; Francisco, S.; Harrison, R. E.; Heuser, J. S.; Jordan, R. L.; Justus, J.; Manning, B.

1978-01-01

The available and optimal methods for generating SAR imagery for NASA applications were identified. The SAR image quality and data processing requirements associated with these applications were studied. Mathematical operations and algorithms required to process sensor data into SAR imagery were defined. The architecture of SAR image formation processors was discussed, and technology necessary to implement the SAR data processors used in both general purpose and dedicated imaging systems was addressed.

High speed optical object recognition processor with massive holographic memory

NASA Technical Reports Server (NTRS)

Chao, T.; Zhou, H.; Reyes, G.

2002-01-01

Real-time object recognition using a compact grayscale optical correlator will be introduced. A holographic memory module for storing a large bank of optimum correlation filters, to accommodate the large data throughput rate needed for many real-world applications, has also been developed. System architecture of the optical processor and the holographic memory will be presented. Application examples of this object recognition technology will also be demonstrated.

The Advanced Communication Technology Satellite and ISDN

NASA Technical Reports Server (NTRS)

Lowry, Peter A.

1996-01-01

This paper depicts the Advanced Communication Technology Satellite (ACTS) system as a global central office switch. The ground portion of the system is the collection of earth stations or T1-VSAT's (T1 very small aperture terminals). The control software for the T1-VSAT's resides in a single CPU. The software consists of two modules, the modem manager and the call manager. The modem manager (MM) controls the RF modem portion of the T1-VSAT. It processes the orderwires from the satellite or from signaling generated by the call manager (CM). The CM controls the Recom Laboratories MSPs by receiving signaling messages from the stacked MSP shelves ro units and sending appropriate setup commands to them. There are two methods used to setup and process calls in the CM; first by dialing up a circuit using a standard telephone handset or, secondly by using an external processor connected to the CPU's second COM port, by sending and receiving signaling orderwires. It is the use of the external processor which permits the ISDN (Integrated Services Digital Network) Signaling Processor to implement ISDN calls. In August 1993, the initial testing of the ISDN Signaling Processor was carried out at ACTS System Test at Lockheed Marietta, Princeton, NJ using the spacecraft in its test configuration on the ground.

FPGA implementation of ICA algorithm for blind signal separation and adaptive noise canceling.

PubMed

Kim, Chang-Min; Park, Hyung-Min; Kim, Taesu; Choi, Yoon-Kyung; Lee, Soo-Young

2003-01-01

An field programmable gate array (FPGA) implementation of independent component analysis (ICA) algorithm is reported for blind signal separation (BSS) and adaptive noise canceling (ANC) in real time. In order to provide enormous computing power for ICA-based algorithms with multipath reverberation, a special digital processor is designed and implemented in FPGA. The chip design fully utilizes modular concept and several chips may be put together for complex applications with a large number of noise sources. Experimental results with a fabricated test board are reported for ANC only, BSS only, and simultaneous ANC/BSS, which demonstrates successful speech enhancement in real environments in real time.

Nonlinear Frequency Compression in Hearing Aids: Impact on Speech and Language Development

PubMed Central

Bentler, Ruth; Walker, Elizabeth; McCreery, Ryan; Arenas, Richard M.; Roush, Patricia

2015-01-01

Objectives The research questions of this study were: (1) Are children using nonlinear frequency compression (NLFC) in their hearing aids getting better access to the speech signal than children using conventional processing schemes? The authors hypothesized that children whose hearing aids provided wider input bandwidth would have more access to the speech signal, as measured by an adaptation of the Speech Intelligibility Index, and (2) are speech and language skills different for children who have been fit with the two different technologies; if so, in what areas? The authors hypothesized that if the children were getting increased access to the speech signal as a result of their NLFC hearing aids (question 1), it would be possible to see improved performance in areas of speech production, morphosyntax, and speech perception compared with the group with conventional processing. Design Participants included 66 children with hearing loss recruited as part of a larger multisite National Institutes of Health–funded study, Outcomes for Children with Hearing Loss, designed to explore the developmental outcomes of children with mild to severe hearing loss. For the larger study, data on communication, academic and psychosocial skills were gathered in an accelerated longitudinal design, with entry into the study between 6 months and 7 years of age. Subjects in this report consisted of 3-, 4-, and 5-year-old children recruited at the North Carolina test site. All had at least at least 6 months of current hearing aid usage with their NLFC or conventional amplification. Demographic characteristics were compared at the three age levels as well as audibility and speech/language outcomes; speech-perception scores were compared for the 5-year-old groups. Results Results indicate that the audibility provided did not differ between the technology options. As a result, there was no difference between groups on speech or language outcome measures at 4 or 5 years of age, and no impact on speech perception (measured at 5 years of age). The difference in Comprehensive Assessment of Spoken Language and mean length of utterance scores for the 3-year-old group favoring the group with conventional amplification may be a consequence of confounding factors such as increased incidence of prematurity in the group using NLFC. Conclusions Children fit with NLFC had similar audibility, as measured by a modified Speech Intelligibility Index, compared with a matched group of children using conventional technology. In turn, there were no differences in their speech and language abilities. PMID:24892229

Nonlinear frequency compression in hearing aids: impact on speech and language development.

PubMed

Bentler, Ruth; Walker, Elizabeth; McCreery, Ryan; Arenas, Richard M; Roush, Patricia

2014-01-01

The research questions of this study were: (1) Are children using nonlinear frequency compression (NLFC) in their hearing aids getting better access to the speech signal than children using conventional processing schemes? The authors hypothesized that children whose hearing aids provided wider input bandwidth would have more access to the speech signal, as measured by an adaptation of the Speech Intelligibility Index, and (2) are speech and language skills different for children who have been fit with the two different technologies; if so, in what areas? The authors hypothesized that if the children were getting increased access to the speech signal as a result of their NLFC hearing aids (question 1), it would be possible to see improved performance in areas of speech production, morphosyntax, and speech perception compared with the group with conventional processing. Participants included 66 children with hearing loss recruited as part of a larger multisite National Institutes of Health-funded study, Outcomes for Children with Hearing Loss, designed to explore the developmental outcomes of children with mild to severe hearing loss. For the larger study, data on communication, academic and psychosocial skills were gathered in an accelerated longitudinal design, with entry into the study between 6 months and 7 years of age. Subjects in this report consisted of 3-, 4-, and 5-year-old children recruited at the North Carolina test site. All had at least at least 6 months of current hearing aid usage with their NLFC or conventional amplification. Demographic characteristics were compared at the three age levels as well as audibility and speech/language outcomes; speech-perception scores were compared for the 5-year-old groups. Results indicate that the audibility provided did not differ between the technology options. As a result, there was no difference between groups on speech or language outcome measures at 4 or 5 years of age, and no impact on speech perception (measured at 5 years of age). The difference in Comprehensive Assessment of Spoken Language and mean length of utterance scores for the 3-year-old group favoring the group with conventional amplification may be a consequence of confounding factors such as increased incidence of prematurity in the group using NLFC. Children fit with NLFC had similar audibility, as measured by a modified Speech Intelligibility Index, compared with a matched group of children using conventional technology. In turn, there were no differences in their speech and language abilities.

The evolutionary development of high specific impulse electric thruster technology

NASA Technical Reports Server (NTRS)

Sovey, James S.; Hamley, John A.; Patterson, Michael J.; Rawlin, Vincent K.; Myers, Roger M.

1992-01-01

Electric propulsion flight and technology demonstrations conducted primarily by Europe, Japan, China, the U.S., and the USSR are reviewed. Evolutionary mission applications for high specific impulse electric thruster systems are discussed, and the status of arcjet, ion, and magnetoplasmadynamic thrusters and associated power processor technologies are summarized.

SDI Software Technology Program Plan Version 1.5

DTIC Science & Technology

1987-06-01

computer generation of auditory communication of meaningful speech. Most speech synthesizers are based on mathematical models of the human vocal tract, but...oral/ auditory and multimodal communications. Although such state-of-the-art interaction technology has not fully matured, user experience has...superior I pattern matching capabilities and the subliminal intuitive deduction capability. The error performance of humans can be helped by careful

Measuring listening effort: driving simulator vs. simple dual-task paradigm

PubMed Central

Wu, Yu-Hsiang; Aksan, Nazan; Rizzo, Matthew; Stangl, Elizabeth; Zhang, Xuyang; Bentler, Ruth

2014-01-01

Objectives The dual-task paradigm has been widely used to measure listening effort. The primary objectives of the study were to (1) investigate the effect of hearing aid amplification and a hearing aid directional technology on listening effort measured by a complicated, more real world dual-task paradigm, and (2) compare the results obtained with this paradigm to a simpler laboratory-style dual-task paradigm. Design The listening effort of adults with hearing impairment was measured using two dual-task paradigms, wherein participants performed a speech recognition task simultaneously with either a driving task in a simulator or a visual reaction-time task in a sound-treated booth. The speech materials and road noises for the speech recognition task were recorded in a van traveling on the highway in three hearing aid conditions: unaided, aided with omni directional processing (OMNI), and aided with directional processing (DIR). The change in the driving task or the visual reaction-time task performance across the conditions quantified the change in listening effort. Results Compared to the driving-only condition, driving performance declined significantly with the addition of the speech recognition task. Although the speech recognition score was higher in the OMNI and DIR conditions than in the unaided condition, driving performance was similar across these three conditions, suggesting that listening effort was not affected by amplification and directional processing. Results from the simple dual-task paradigm showed a similar trend: hearing aid technologies improved speech recognition performance, but did not affect performance in the visual reaction-time task (i.e., reduce listening effort). The correlation between listening effort measured using the driving paradigm and the visual reaction-time task paradigm was significant. The finding showing that our older (56 to 85 years old) participants’ better speech recognition performance did not result in reduced listening effort was not consistent with literature that evaluated younger (approximately 20 years old), normal hearing adults. Because of this, a follow-up study was conducted. In the follow-up study, the visual reaction-time dual-task experiment using the same speech materials and road noises was repeated on younger adults with normal hearing. Contrary to findings with older participants, the results indicated that the directional technology significantly improved performance in both speech recognition and visual reaction-time tasks. Conclusions Adding a speech listening task to driving undermined driving performance. Hearing aid technologies significantly improved speech recognition while driving, but did not significantly reduce listening effort. Listening effort measured by dual-task experiments using a simulated real-world driving task and a conventional laboratory-style task was generally consistent. For a given listening environment, the benefit of hearing aid technologies on listening effort measured from younger adults with normal hearing may not be fully translated to older listeners with hearing impairment. PMID:25083599

Implicit theory manipulations affecting efficacy of a smartphone application aiding speech therapy for Parkinson's patients.

PubMed

Nolan, Peter; Hoskins, Sherria; Johnson, Julia; Powell, Vaughan; Chaudhuri, K Ray; Eglin, Roger

2012-01-01

A Smartphone speech-therapy application (STA) is being developed, intended for people with Parkinson's disease (PD) with reduced implicit volume cues. The STA offers visual volume feedback, addressing diminished auditory cues. Users are typically older adults, less familiar with new technology. Domain-specific implicit theories (ITs) have been shown to result in mastery or helpless behaviors. Studies manipulating participants' implicit theories of 'technology' (Study One), and 'ability to affect one's voice' (Study Two), were coordinated with iterative STA test-stages, using patients with PD with prior speech-therapist referrals. Across studies, findings suggest it is possible to manipulate patients' ITs related to engaging with a Smartphone STA. This potentially impacts initial application approach and overall effort using a technology-based therapy.

A Survey of Techniques for Modeling and Improving Reliability of Computing Systems

DOE PAGES

Mittal, Sparsh; Vetter, Jeffrey S.

2015-04-24

Recent trends of aggressive technology scaling have greatly exacerbated the occurrences and impact of faults in computing systems. This has made `reliability' a first-order design constraint. To address the challenges of reliability, several techniques have been proposed. In this study, we provide a survey of architectural techniques for improving resilience of computing systems. We especially focus on techniques proposed for microarchitectural components, such as processor registers, functional units, cache and main memory etc. In addition, we discuss techniques proposed for non-volatile memory, GPUs and 3D-stacked processors. To underscore the similarities and differences of the techniques, we classify them based onmore » their key characteristics. We also review the metrics proposed to quantify vulnerability of processor structures. Finally, we believe that this survey will help researchers, system-architects and processor designers in gaining insights into the techniques for improving reliability of computing systems.« less

A GaAs vector processor based on parallel RISC microprocessors

NASA Astrophysics Data System (ADS)

Misko, Tim A.; Rasset, Terry L.

A vector processor architecture based on the development of a 32-bit microprocessor using gallium arsenide (GaAs) technology has been developed. The McDonnell Douglas vector processor (MVP) will be fabricated completely from GaAs digital integrated circuits. The MVP architecture includes a vector memory of 1 megabyte, a parallel bus architecture with eight processing elements connected in parallel, and a control processor. The processing elements consist of a reduced instruction set CPU (RISC) with four floating-point coprocessor units and necessary memory interface functions. This architecture has been simulated for several benchmark programs including complex fast Fourier transform (FFT), complex inner product, trigonometric functions, and sort-merge routine. The results of this study indicate that the MVP can process a 1024-point complex FFT at a speed of 112 microsec (389 megaflops) while consuming approximately 618 W of power in a volume of approximately 0.1 ft-cubed.

A Survey of Techniques for Modeling and Improving Reliability of Computing Systems

DOE Office of Scientific and Technical Information (OSTI.GOV)

Mittal, Sparsh; Vetter, Jeffrey S.

Recent trends of aggressive technology scaling have greatly exacerbated the occurrences and impact of faults in computing systems. This has made `reliability' a first-order design constraint. To address the challenges of reliability, several techniques have been proposed. In this study, we provide a survey of architectural techniques for improving resilience of computing systems. We especially focus on techniques proposed for microarchitectural components, such as processor registers, functional units, cache and main memory etc. In addition, we discuss techniques proposed for non-volatile memory, GPUs and 3D-stacked processors. To underscore the similarities and differences of the techniques, we classify them based onmore » their key characteristics. We also review the metrics proposed to quantify vulnerability of processor structures. Finally, we believe that this survey will help researchers, system-architects and processor designers in gaining insights into the techniques for improving reliability of computing systems.« less

Long-Term Outcomes of Speech Therapy for Seven Adolescents with Visual Feedback Technologies: Ultrasound and Electropalatography

ERIC Educational Resources Information Center

Bacsfalvi, Penelope; Bernhardt, Barbara May

2011-01-01

This follow-up study investigated the speech production of seven adolescents and young adults with hearing impairment 2-4 years after speech intervention with ultrasound and electropalatography. Perceptual judgments by seven expert listeners revealed that five out of seven speakers either continued to generalize post-treatment or maintained their…

Ultrasound as Visual Feedback in Speech Habilitation: Exploring Consultative Use in Rural British Columbia, Canada

ERIC Educational Resources Information Center

Bernhardt, B. May; Bacsfalvi, Penelope; Adler-Bock, Marcy; Shimizu, Reiko; Cheney, Audrey; Giesbrecht, Nathan; O'Connell, Maureen; Sirianni, Jason; Radanov, Bosko

2008-01-01

Ultrasound has shown promise as a visual feedback tool in speech therapy. Rural clients, however, often have minimal access to new technologies. The purpose of the current study was to evaluate consultative treatment using ultrasound in rural communities. Two speech-language pathologists (SLPs) trained in ultrasound use provided consultation with…

Implicit prosody mining based on the human eye image capture technology

NASA Astrophysics Data System (ADS)

Gao, Pei-pei; Liu, Feng

2013-08-01

The technology of eye tracker has become the main methods of analyzing the recognition issues in human-computer interaction. Human eye image capture is the key problem of the eye tracking. Based on further research, a new human-computer interaction method introduced to enrich the form of speech synthetic. We propose a method of Implicit Prosody mining based on the human eye image capture technology to extract the parameters from the image of human eyes when reading, control and drive prosody generation in speech synthesis, and establish prosodic model with high simulation accuracy. Duration model is key issues for prosody generation. For the duration model, this paper put forward a new idea for obtaining gaze duration of eyes when reading based on the eye image capture technology, and synchronous controlling this duration and pronunciation duration in speech synthesis. The movement of human eyes during reading is a comprehensive multi-factor interactive process, such as gaze, twitching and backsight. Therefore, how to extract the appropriate information from the image of human eyes need to be considered and the gaze regularity of eyes need to be obtained as references of modeling. Based on the analysis of current three kinds of eye movement control model and the characteristics of the Implicit Prosody reading, relative independence between speech processing system of text and eye movement control system was discussed. It was proved that under the same text familiarity condition, gaze duration of eyes when reading and internal voice pronunciation duration are synchronous. The eye gaze duration model based on the Chinese language level prosodic structure was presented to change previous methods of machine learning and probability forecasting, obtain readers' real internal reading rhythm and to synthesize voice with personalized rhythm. This research will enrich human-computer interactive form, and will be practical significance and application prospect in terms of disabled assisted speech interaction. Experiments show that Implicit Prosody mining based on the human eye image capture technology makes the synthesized speech has more flexible expressions.

Challenges and Recent Developments in Hearing Aids: Part I. Speech Understanding in Noise, Microphone Technologies and Noise Reduction Algorithms

PubMed Central

Chung, King

2004-01-01

This review discusses the challenges in hearing aid design and fitting and the recent developments in advanced signal processing technologies to meet these challenges. The first part of the review discusses the basic concepts and the building blocks of digital signal processing algorithms, namely, the signal detection and analysis unit, the decision rules, and the time constants involved in the execution of the decision. In addition, mechanisms and the differences in the implementation of various strategies used to reduce the negative effects of noise are discussed. These technologies include the microphone technologies that take advantage of the spatial differences between speech and noise and the noise reduction algorithms that take advantage of the spectral difference and temporal separation between speech and noise. The specific technologies discussed in this paper include first-order directional microphones, adaptive directional microphones, second-order directional microphones, microphone matching algorithms, array microphones, multichannel adaptive noise reduction algorithms, and synchrony detection noise reduction algorithms. Verification data for these technologies, if available, are also summarized. PMID:15678225

Speech perception and production in severe environments

NASA Astrophysics Data System (ADS)

Pisoni, David B.

1990-09-01

The goal was to acquire new knowledge about speech perception and production in severe environments such as high masking noise, increased cognitive load or sustained attentional demands. Changes were examined in speech production under these adverse conditions through acoustic analysis techniques. One set of studies focused on the effects of noise on speech production. The experiments in this group were designed to generate a database of speech obtained in noise and in quiet. A second set of experiments was designed to examine the effects of cognitive load on the acoustic-phonetic properties of speech. Talkers were required to carry out a demanding perceptual motor task while they read lists of test words. A final set of experiments explored the effects of vocal fatigue on the acoustic-phonetic properties of speech. Both cognitive load and vocal fatigue are present in many applications where speech recognition technology is used, yet their influence on speech production is poorly understood.

An integrated approach to improving noisy speech perception

NASA Astrophysics Data System (ADS)

Koval, Serguei; Stolbov, Mikhail; Smirnova, Natalia; Khitrov, Mikhail

2002-05-01

For a number of practical purposes and tasks, experts have to decode speech recordings of very poor quality. A combination of techniques is proposed to improve intelligibility and quality of distorted speech messages and thus facilitate their comprehension. Along with the application of noise cancellation and speech signal enhancement techniques removing and/or reducing various kinds of distortions and interference (primarily unmasking and normalization in time and frequency fields), the approach incorporates optimal listener expert tactics based on selective listening, nonstandard binaural listening, accounting for short-term and long-term human ear adaptation to noisy speech, as well as some methods of speech signal enhancement to support speech decoding during listening. The approach integrating the suggested techniques ensures high-quality ultimate results and has successfully been applied by Speech Technology Center experts and by numerous other users, mainly forensic institutions, to perform noisy speech records decoding for courts, law enforcement and emergency services, accident investigation bodies, etc.

Onboard Interferometric SAR Processor for the Ka-Band Radar Interferometer (KaRIn)

NASA Technical Reports Server (NTRS)

Esteban-Fernandez, Daniel; Rodriquez, Ernesto; Peral, Eva; Clark, Duane I.; Wu, Xiaoqing

2011-01-01

An interferometric synthetic aperture radar (SAR) onboard processor concept and algorithm has been developed for the Ka-band radar interferometer (KaRIn) instrument on the Surface and Ocean Topography (SWOT) mission. This is a mission- critical subsystem that will perform interferometric SAR processing and multi-look averaging over the oceans to decrease the data rate by three orders of magnitude, and therefore enable the downlink of the radar data to the ground. The onboard processor performs demodulation, range compression, coregistration, and re-sampling, and forms nine azimuth squinted beams. For each of them, an interferogram is generated, including common-band spectral filtering to improve correlation, followed by averaging to the final 1 1-km ground resolution pixel. The onboard processor has been prototyped on a custom FPGA-based cPCI board, which will be part of the radar s digital subsystem. The level of complexity of this technology, dictated by the implementation of interferometric SAR processing at high resolution, the extremely tight level of accuracy required, and its implementation on FPGAs are unprecedented at the time of this reporting for an onboard processor for flight applications.

Telerehabilitation, virtual therapists, and acquired neurologic speech and language disorders.

PubMed

Cherney, Leora R; van Vuuren, Sarel

2012-08-01

Telerehabilitation (telerehab) offers cost-effective services that potentially can improve access to care for those with acquired neurologic communication disorders. However, regulatory issues including licensure, reimbursement, and threats to privacy and confidentiality hinder the routine implementation of telerehab services into the clinical setting. Despite these barriers, rapid technological advances and a growing body of research regarding the use of telerehab applications support its use. This article reviews the evidence related to acquired neurologic speech and language disorders in adults, focusing on studies that have been published since 2000. Research studies have used telerehab systems to assess and treat disorders including dysarthria, apraxia of speech, aphasia, and mild Alzheimer disease. They show that telerehab is a valid and reliable vehicle for delivering speech and language services. The studies represent a progression of technological advances in computing, Internet, and mobile technologies. They range on a continuum from working synchronously (in real-time) with a speech-language pathologist to working asynchronously (offline) with a stand-in virtual therapist. One such system that uses a virtual therapist for the treatment of aphasia, the Web-ORLA™ (Rehabilitation Institute of Chicago, Chicago, IL) system, is described in detail. Future directions for the advancement of telerehab for clinical practice are discussed. Thieme Medical Publishers 333 Seventh Avenue, New York, NY 10001, USA.

Exploring expressivity and emotion with artificial voice and speech technologies.

PubMed

Pauletto, Sandra; Balentine, Bruce; Pidcock, Chris; Jones, Kevin; Bottaci, Leonardo; Aretoulaki, Maria; Wells, Jez; Mundy, Darren P; Balentine, James

2013-10-01

Emotion in audio-voice signals, as synthesized by text-to-speech (TTS) technologies, was investigated to formulate a theory of expression for user interface design. Emotional parameters were specified with markup tags, and the resulting audio was further modulated with post-processing techniques. Software was then developed to link a selected TTS synthesizer with an automatic speech recognition (ASR) engine, producing a chatbot that could speak and listen. Using these two artificial voice subsystems, investigators explored both artistic and psychological implications of artificial speech emotion. Goals of the investigation were interdisciplinary, with interest in musical composition, augmentative and alternative communication (AAC), commercial voice announcement applications, human-computer interaction (HCI), and artificial intelligence (AI). The work-in-progress points towards an emerging interdisciplinary ontology for artificial voices. As one study output, HCI tools are proposed for future collaboration.

Speech-Language Pathologists' Perceptions of the Importance and Ability to Use Assistive Technology in the Kingdom of Saudi Arabia

ERIC Educational Resources Information Center

Al-Dawaideh, Ahmad Mousa

2013-01-01

Speech-language pathologists (SLPs) frequently work with people with severe communication disorders who require assistive technology (AT) for communication. The purpose of this study was to investigate the SLPs perceptions of the importance of and ability level required for using AT, and the relationship of AT with gender, level of education,…

Music therapy as specific and complementary training for adults after cochlear implantation: A pilot study.

PubMed

Hutter, E; Argstatter, H; Grapp, M; Plinkert, P K

2015-09-01

Although cochlear implant (CI) users achieve good speech comprehension, they experience difficulty perceiving music and prosody in speech. As the provision of music training in rehabilitation is limited, a novel concept of music therapy for rehabilitation of adult CI users was developed and evaluated in this pilot study. Twelve unilaterally implanted, postlingually deafened CI users attended ten sessions of individualized and standardized training. The training started about 6 weeks after the initial activation of the speech processor. Before and after therapy, psychological and musical tests were applied in order to evaluate the effects of music therapy. CI users completed the musical tests in two conditions: bilateral (CI + contralateral, unimplanted ear) and unilateral (CI only). After therapy, improvements were observed in the subjective sound quality (Hearing Implant Sound Quality Index) and the global score on the self-concept questionnaire (Multidimensional Self-Concept Scales) as well as in the musical subtests for melody recognition and for timbre identification in the unilateral condition. Discussion Preliminary results suggest improvements in subjective hearing and music perception, with an additional increase in global self-concept and enhanced daily listening capacities. The novel concept of individualized music therapy seems to provide an effective treatment option in the rehabilitation of adult CI users. Further investigations are necessary to evaluate effects in the area of prosody perception and to separate therapy effects from general learning effects in CI rehabilitation.

Effects of Compression on Speech Acoustics, Intelligibility, and Sound Quality

PubMed Central

Souza, Pamela E.

2002-01-01

The topic of compression has been discussed quite extensively in the last 20 years (eg, Braida et al., 1982; Dillon, 1996, 2000; Dreschler, 1992; Hickson, 1994; Kuk, 2000 and 2002; Kuk and Ludvigsen, 1999; Moore, 1990; Van Tasell, 1993; Venema, 2000; Verschuure et al., 1996; Walker and Dillon, 1982). However, the latest comprehensive update by this journal was published in 1996 (Kuk, 1996). Since that time, use of compression hearing aids has increased dramatically, from half of hearing aids dispensed only 5 years ago to four out of five hearing aids dispensed today (Strom, 2002b). Most of today's digital and digitally programmable hearing aids are compression devices (Strom, 2002a). It is probable that within a few years, very few patients will be fit with linear hearing aids. Furthermore, compression has increased in complexity, with greater numbers of parameters under the clinician's control. Ideally, these changes will translate to greater flexibility and precision in fitting and selection. However, they also increase the need for information about the effects of compression amplification on speech perception and speech quality. As evidenced by the large number of sessions at professional conferences on fitting compression hearing aids, clinicians continue to have questions about compression technology and when and how it should be used. How does compression work? Who are the best candidates for this technology? How should adjustable parameters be set to provide optimal speech recognition? What effect will compression have on speech quality? These and other questions continue to drive our interest in this technology. This article reviews the effects of compression on the speech signal and the implications for speech intelligibility, quality, and design of clinical procedures. PMID:25425919

Cortical activation with sound stimulation in cochlear implant users demonstrated by positron emission tomography.

PubMed

Naito, Y; Okazawa, H; Honjo, I; Hirano, S; Takahashi, H; Shiomi, Y; Hoji, W; Kawano, M; Ishizu, K; Yonekura, Y

1995-07-01

Six postlingually deaf patients using multi-channel cochlear implants were examined by positron emission tomography (PET) using 15O-labeled water. Changes in regional cerebral blood flow (rCBF) were measured during different sound stimuli. The stimulation paradigms employed consisted of two sets of three different conditions; (1) no sound stimulation with the speech processor of the cochlear implant system switched off, (2) hearing white noise and (3) hearing sequential Japanese sentences. In the primary auditory area, the mean rCBF increase during noise stimulation was significantly greater on the side contralateral to the implant than on the ipsilateral side. Speech stimulation caused significantly greater rCBF increase compared with noise stimulation in the left immediate auditory association area (P < 0.01), the bilateral auditory association areas (P < 0.01), the posterior part of the bilateral inferior frontal gyri; the Broca's area (P < 0.01) and its right hemisphere homologue (P < 0.05). Activation of cortices related to verbal and non-verbal sound recognition was clearly demonstrated in the current subjects probably because complete silence was attained in the control condition.

49 CFR 234.275 - Processor-based systems.

Code of Federal Regulations, 2011 CFR

2011-10-01

... new or novel technology, or which provide safety-critical data to a railroad signal or train control... requirements. New or novel technology refers to a technology not previously recognized for use as of March 7... but which provides safety-critical data to a signal or train control system shall be included in the...

49 CFR 234.275 - Processor-based systems.

Code of Federal Regulations, 2010 CFR

2010-10-01

... new or novel technology, or which provide safety-critical data to a railroad signal or train control... requirements. New or novel technology refers to a technology not previously recognized for use as of March 7... but which provides safety-critical data to a signal or train control system shall be included in the...

Comparison of speech recognition with adaptive digital and FM remote microphone hearing assistance technology by listeners who use hearing aids.

PubMed

Thibodeau, Linda

2014-06-01

The purpose of this study was to compare the benefits of 3 types of remote microphone hearing assistance technology (HAT), adaptive digital broadband, adaptive frequency modulation (FM), and fixed FM, through objective and subjective measures of speech recognition in clinical and real-world settings. Participants included 11 adults, ages 16 to 78 years, with primarily moderate-to-severe bilateral hearing impairment (HI), who wore binaural behind-the-ear hearing aids; and 15 adults, ages 18 to 30 years, with normal hearing. Sentence recognition in quiet and in noise and subjective ratings were obtained in 3 conditions of wireless signal processing. Performance by the listeners with HI when using the adaptive digital technology was significantly better than that obtained with the FM technology, with the greatest benefits at the highest noise levels. The majority of listeners also preferred the digital technology when listening in a real-world noisy environment. The wireless technology allowed persons with HI to surpass persons with normal hearing in speech recognition in noise, with the greatest benefit occurring with adaptive digital technology. The use of adaptive digital technology combined with speechreading cues would allow persons with HI to engage in communication in environments that would have otherwise not been possible with traditional wireless technology.

Public Speaking Anxiety: Comparing Face-to-Face and Web-Based Speeches

ERIC Educational Resources Information Center

Campbell, Scott; Larson, James

2013-01-01

This study is to determine whether or not students have a different level of anxiety between giving a speech to a group of people in a traditional face-to-face classroom setting to a speech given to an audience (visible on a projected screen) into a camera using distance or web-based technology. The study included approximately 70 students.…

Assessing Children's Home Language Environments Using Automatic Speech Recognition Technology

ERIC Educational Resources Information Center

Greenwood, Charles R.; Thiemann-Bourque, Kathy; Walker, Dale; Buzhardt, Jay; Gilkerson, Jill

2011-01-01

The purpose of this research was to replicate and extend some of the findings of Hart and Risley using automatic speech processing instead of human transcription of language samples. The long-term goal of this work is to make the current approach to speech processing possible by researchers and clinicians working on a daily basis with families and…

An Exploration of the Potential of Automatic Speech Recognition to Assist and Enable Receptive Communication in Higher Education

ERIC Educational Resources Information Center

Wald, Mike

2006-01-01

The potential use of Automatic Speech Recognition to assist receptive communication is explored. The opportunities and challenges that this technology presents students and staff to provide captioning of speech online or in classrooms for deaf or hard of hearing students and assist blind, visually impaired or dyslexic learners to read and search…

Emerging technologies with potential for objectively evaluating speech recognition skills.

PubMed

Rawool, Vishakha Waman

2016-01-01

Work-related exposure to noise and other ototoxins can cause damage to the cochlea, synapses between the inner hair cells, the auditory nerve fibers, and higher auditory pathways, leading to difficulties in recognizing speech. Procedures designed to determine speech recognition scores (SRS) in an objective manner can be helpful in disability compensation cases where the worker claims to have poor speech perception due to exposure to noise or ototoxins. Such measures can also be helpful in determining SRS in individuals who cannot provide reliable responses to speech stimuli, including patients with Alzheimer's disease, traumatic brain injuries, and infants with and without hearing loss. Cost-effective neural monitoring hardware and software is being rapidly refined due to the high demand for neurogaming (games involving the use of brain-computer interfaces), health, and other applications. More specifically, two related advances in neuro-technology include relative ease in recording neural activity and availability of sophisticated analysing techniques. These techniques are reviewed in the current article and their applications for developing objective SRS procedures are proposed. Issues related to neuroaudioethics (ethics related to collection of neural data evoked by auditory stimuli including speech) and neurosecurity (preservation of a person's neural mechanisms and free will) are also discussed.

Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

NASA Technical Reports Server (NTRS)

Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

2010-01-01

A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.

Educational Applications for Blind and Partially Sighted Pupils Based on Speech Technologies for Serbian.

PubMed

Lučić, Branko; Ostrogonac, Stevan; Vujnović Sedlar, Nataša; Sečujski, Milan

2015-01-01

The inclusion of persons with disabilities has always represented an important issue. Advancements within the field of computer science have enabled the development of different types of aids, which have significantly improved the quality of life of the disabled. However, for some disabilities, such as visual impairment, the purpose of these aids is to establish an alternative communication channel and thus overcome the user's disability. Speech technologies play the crucial role in this process. This paper presents the ongoing efforts to create a set of educational applications based on speech technologies for Serbian for the early stages of education of blind and partially sighted children. Two educational applications dealing with memory exercises and comprehension of geometrical shapes are presented, along with the initial tests results obtained from research including visually impaired pupils.

Educational Applications for Blind and Partially Sighted Pupils Based on Speech Technologies for Serbian

PubMed Central

Lučić, Branko; Ostrogonac, Stevan; Vujnović Sedlar, Nataša; Sečujski, Milan

2015-01-01

The inclusion of persons with disabilities has always represented an important issue. Advancements within the field of computer science have enabled the development of different types of aids, which have significantly improved the quality of life of the disabled. However, for some disabilities, such as visual impairment, the purpose of these aids is to establish an alternative communication channel and thus overcome the user's disability. Speech technologies play the crucial role in this process. This paper presents the ongoing efforts to create a set of educational applications based on speech technologies for Serbian for the early stages of education of blind and partially sighted children. Two educational applications dealing with memory exercises and comprehension of geometrical shapes are presented, along with the initial tests results obtained from research including visually impaired pupils. PMID:26171422

Speech Recognition as a Transcription Aid: A Randomized Comparison With Standard Transcription

PubMed Central

Mohr, David N.; Turner, David W.; Pond, Gregory R.; Kamath, Joseph S.; De Vos, Cathy B.; Carpenter, Paul C.

2003-01-01

Objective. Speech recognition promises to reduce information entry costs for clinical information systems. It is most likely to be accepted across an organization if physicians can dictate without concerning themselves with real-time recognition and editing; assistants can then edit and process the computer-generated document. Our objective was to evaluate the use of speech-recognition technology in a randomized controlled trial using our institutional infrastructure. Design. Clinical note dictations from physicians in two specialty divisions were randomized to either a standard transcription process or a speech-recognition process. Secretaries and transcriptionists also were assigned randomly to each of these processes. Measurements. The duration of each dictation was measured. The amount of time spent processing a dictation to yield a finished document also was measured. Secretarial and transcriptionist productivity, defined as hours of secretary work per minute of dictation processed, was determined for speech recognition and standard transcription. Results. Secretaries in the endocrinology division were 87.3% (confidence interval, 83.3%, 92.3%) as productive with the speech-recognition technology as implemented in this study as they were using standard transcription. Psychiatry transcriptionists and secretaries were similarly less productive. Author, secretary, and type of clinical note were significant (p < 0.05) predictors of productivity. Conclusion. When implemented in an organization with an existing document-processing infrastructure (which included training and interfaces of the speech-recognition editor with the existing document entry application), speech recognition did not improve the productivity of secretaries or transcriptionists. PMID:12509359

A real-time phoneme counting algorithm and application for speech rate monitoring.

PubMed

Aharonson, Vered; Aharonson, Eran; Raichlin-Levi, Katia; Sotzianu, Aviv; Amir, Ofer; Ovadia-Blechman, Zehava

2017-03-01

Adults who stutter can learn to control and improve their speech fluency by modifying their speaking rate. Existing speech therapy technologies can assist this practice by monitoring speaking rate and providing feedback to the patient, but cannot provide an accurate, quantitative measurement of speaking rate. Moreover, most technologies are too complex and costly to be used for home practice. We developed an algorithm and a smartphone application that monitor a patient's speaking rate in real time and provide user-friendly feedback to both patient and therapist. Our speaking rate computation is performed by a phoneme counting algorithm which implements spectral transition measure extraction to estimate phoneme boundaries. The algorithm is implemented in real time in a mobile application that presents its results in a user-friendly interface. The application incorporates two modes: one provides the patient with visual feedback of his/her speech rate for self-practice and another provides the speech therapist with recordings, speech rate analysis and tools to manage the patient's practice. The algorithm's phoneme counting accuracy was validated on ten healthy subjects who read a paragraph at slow, normal and fast paces, and was compared to manual counting of speech experts. Test-retest and intra-counter reliability were assessed. Preliminary results indicate differences of -4% to 11% between automatic and human phoneme counting. Differences were largest for slow speech. The application can thus provide reliable, user-friendly, real-time feedback for speaking rate control practice. Copyright © 2017 Elsevier Inc. All rights reserved.

"The Communication Needs and Rights of Mankind", Group 1 Report of the Futuristic Priorities Division of the Speech Communication Association. "Future Communication Technologies; Hardware and Software"; Group 2 Report.

ERIC Educational Resources Information Center

Dance, Frank E. X.; And Others

This paper reports on the Futuristic Priorities Division members' recommendations and priorities concerning the impact of the future on communication and on the speech communication discipline. The recommendations and priorities are listed for two subgroups: The Communication Needs and Rights of Mankind; and Future Communication Technologies:…

Phonetics and Technology in the Classroom: A Practical Approach to Using Speech Analysis Software in Second-Language Pronunciation Instruction

ERIC Educational Resources Information Center

Olsen, Daniel J.

2014-01-01

While speech analysis technology has become an integral part of phonetic research, and to some degree is used in language instruction at the most advanced levels, it appears to be mostly absent from the beginning levels of language instruction. In part, the lack of incorporation into the language classroom can be attributed to both the lack of…

A Pilot Investigation regarding Speech-Recognition Performance in Noise for Adults with Hearing Loss in the FM+HA Listening Condition

ERIC Educational Resources Information Center

Lewis, M. Samantha; Gallun, Frederick J.; Gordon, Jane; Lilly, David J.; Crandell, Carl

2010-01-01

While the concurrent use of the hearing aid (HA) microphone with frequency modulation (FM) technology can decrease speech-recognition performance, the FM+HA condition is still an important setting for users of both HA and FM technology. The primary goal of this investigation was to evaluate the effect of attenuating HA gain in the FM+HA listening…

Artificial Intelligence and Its Use in Cost Type analyses with an Example in Cost Performance Measurement.

DTIC Science & Technology

1985-01-01

7-Ai6i 817 ARTIFICIAL INTELLIGENCE AND ITS USE IN COST TYE1/I ANALYSES WdITH ANt EXAMPLE IN COST PERFORMANCE I MERSUREMENT(U) DEFENSE SYSTEMS...INTELLIGENCE-THE EMERGING TECHNOLOGY/ NATURAL LANGUAGE PROCESSORS K ~ With the advent of ARTIFICAL INTELLEGENCE (AI), we are entering into a new era of...language processor which is commerically available is INTELLECT, by Artifical Intellegence Incorporated, Waltham, Mass. To illustrate what a natural

Emerging Radio and Manet Technology Study: Research Support for a Survey of State-of-the-art Commercial and Military Hardware/Software for Mobile Ad Hoc Networks

DTIC Science & Technology

2014-10-01

44 Table 19: Raspberry Pi Information...boards – These are single board devices targeted to education and embedding, the best known being the Raspberry Pi ; and 3. Development boards – These...popular, as it has high performance processor (perhaps 4 times the power of a Raspberry Pi ) with dual core processors running at 1.6 GHz and the cost is

Uses of DARPA Materials Sciences Technology in DoD Systems.

DTIC Science & Technology

1996-05-01

and Lasers NUMBER: University of Central Florida 4000 Central Florida Blvd. P.O. Box 162700 Orlando, FL 32816-2700 9. S PONSO RIN GMO NITO RING AGENCY...course of the program. These advances were communicated to the industry through seminars and workshops, individual plant and agency visits, videotapes on...1995) • P3 ISAR Radar Processor * Digital Signal Processor for OH-58D helicopter * Motorola building a GaAs IC plant for IRIDIUM 26 GALLIUM ARSENIDE

Use of Field Programmable Gate Array Technology in Future Space Avionics

NASA Technical Reports Server (NTRS)

Ferguson, Roscoe C.; Tate, Robert

2005-01-01

Fulfilling NASA's new vision for space exploration requires the development of sustainable, flexible and fault tolerant spacecraft control systems. The traditional development paradigm consists of the purchase or fabrication of hardware boards with fixed processor and/or Digital Signal Processing (DSP) components interconnected via a standardized bus system. This is followed by the purchase and/or development of software. This paradigm has several disadvantages for the development of systems to support NASA's new vision. Building a system to be fault tolerant increases the complexity and decreases the performance of included software. Standard bus design and conventional implementation produces natural bottlenecks. Configuring hardware components in systems containing common processors and DSPs is difficult initially and expensive or impossible to change later. The existence of Hardware Description Languages (HDLs), the recent increase in performance, density and radiation tolerance of Field Programmable Gate Arrays (FPGAs), and Intellectual Property (IP) Cores provides the technology for reprogrammable Systems on a Chip (SOC). This technology supports a paradigm better suited for NASA's vision. Hardware and software production are melded for more effective development; they can both evolve together over time. Designers incorporating this technology into future avionics can benefit from its flexibility. Systems can be designed with improved fault isolation and tolerance using hardware instead of software. Also, these designs can be protected from obsolescence problems where maintenance is compromised via component and vendor availability.To investigate the flexibility of this technology, the core of the Central Processing Unit and Input/Output Processor of the Space Shuttle AP101S Computer were prototyped in Verilog HDL and synthesized into an Altera Stratix FPGA.

[Cochlear implant in children: rational, indications and cost/efficacy].

PubMed

Martini, A; Bovo, R; Trevisi, P; Forli, F; Berrettini, S

2013-06-01

A cochlear implant (CI) is a partially implanted electronic device that can help to provide a sense of sound and support speech to severely to profoundly hearing impaired patients. It is constituted by an external portion, that usually sits behind the ear and an internal portion surgically placed under the skin. The external components include a microphone connected to a speech processor that selects and arranges sounds pucked up by the microphone. This is connected to a transmitter coil, worn on the side of the head, which transmits data to an internal receiver coil placed under the skin. The received data are delivered to an array of electrodes that are surgically implanted within the cochlea. The primary neural targets of the electrodes are the spiral ganglion cells which innervate fibers of the auditory nerve. When the electrodes are activated by the signal, they send a current along the auditory nerve and auditory pathways to the auditory cortex. Children and adults who are profoundly or severely hearing impaired can be fitted with cochlear implants. According to the Food and Drug Administration, approximately 188,000 people worldwide have received implants. In Italy it is extimated that there are about 6-7000 implanted patients, with an average of 700 CI surgeries per year. Cochlear implantation, followed by intensive postimplantation speech therapy, can help young children to acquire speech, language, and social skills. Early implantation provides exposure to sounds that can be helpful during the critical period when children learn speech and language skills. In 2000, the Food and Drug Administration lowered the age of eligibility to 12 months for one type of CI. With regard to the results after cochlear implantation in relation to early implantation, better linguistic results are reported in children implanted before 12 months of life, even if no sufficient data exist regarding the relation between this advantage and the duration of implant use and how long this advantage persists in the subsequent years. With regard to cochlear implantation in children older than 12 months the studies show better hearing and linguistic results in children implanted at earlier ages. A sensitive period under 24-36 months has been identified over which cochlear implantation is reported to be less effective in terms of improvement in speech and hearing results. With regard to clinical effectiveness of bilateral cochlear implantation, greater benefits from bilateral implants compared to monolateral ones when assessing hearing in quiet and in noise and in sound localization abilities are reported to be present in both case of simultaneous or sequential bilateral implantation. However, with regard to the delay between the surgeries in sequential bilateral implantation, although benefit is reported to be present even after very long delays, on average long delays between surgeries seems to negatively affect the outcome with the second implant. With regard to benefits after cochlear implantation in children with multiple disabilities, benefits in terms of speech perception and communication as well as in quality of the daily life are reported even if benefits are slower and lower in comparison to those generally attained by implanted children without additional disabilities. Regarding the costs/efficacy ratio, the CI is expensive, in particular because of the cost of the high technological device, long life support, but even if healthcare costs are high, the savings in terms of indirect costs and quality of life are important. The CI, in fact, has a positive impact in terms of quality of life.

Discovering Motifs in Biological Sequences Using the Micron Automata Processor.

PubMed

Roy, Indranil; Aluru, Srinivas

2016-01-01

Finding approximately conserved sequences, called motifs, across multiple DNA or protein sequences is an important problem in computational biology. In this paper, we consider the (l, d) motif search problem of identifying one or more motifs of length l present in at least q of the n given sequences, with each occurrence differing from the motif in at most d substitutions. The problem is known to be NP-complete, and the largest solved instance reported to date is (26,11). We propose a novel algorithm for the (l,d) motif search problem using streaming execution over a large set of non-deterministic finite automata (NFA). This solution is designed to take advantage of the micron automata processor, a new technology close to deployment that can simultaneously execute multiple NFA in parallel. We demonstrate the capability for solving much larger instances of the (l, d) motif search problem using the resources available within a single automata processor board, by estimating run-times for problem instances (39,18) and (40,17). The paper serves as a useful guide to solving problems using this new accelerator technology.

[Correlation of auditory-verbal skills in patients with cochlear implants and their evaluation in positone emission tomography (PET)].

PubMed

Łukaszewicz, Zuzanna; Soluch, Paweł; Niemczyk, Kazimierz; Lachowska, Magdalena

2010-06-01

An assumption was taken that in central nervous system (CNS) in patients above 15 years of age there are possible mechanisms of neuronal changes. Those changes allow for reconstruction or formation of natural activation pattern of appropriate brain structures responsible for auditory speech processing. The aim of the study was to observe if there are any dynamic functional changes in central nervous system and their correlation to the auditory-verbal skills of the patients. Nine right-handed patients between 15 and 36 years of age were examined, 6 females and 3 males. All of them were treated with cochlear implantation and are in frequent follow-up in the Department of Otolaryngology at the Medical University of Warsaw due to profound sensorineural hearing loss. In present study the patients were examined within 24 hours after the first fitting of the speech processor of the cochlear implant, and 1 and 2 years subsequently. Combination of performed examinations consisted of: positone emission tomography of the brain, and audiological tests including speech assessment. In the group of patients 4 were postlingually deaf, and 5 were prelinqually deaf. Postlingually deaf patients achieved great improvement of hearing and speech understanding. In their first PET examination very intensive activation of visual cortex V1 and V2 (BA17 and 18) was observed. There was no significant activation in the dominant (left) hemisphere of the brain. In PET examination performed 1 and 2 years after the cochlear implantation no more V1 and V2 activation region was observed. Instead particular regions of the left hemisphere got activated. In prelingually deaf patients no significant changes in central nervous system were noticeable neither in PET nor in speech assessment, although their hearing possibilities improved. Positive correlation was observed between the level of speech understanding, linguistic skills and the activation of appropriate areas of the left hemisphere of the brain in postlingually deaf patients treated with cochlear implants. No such correlation was noted in prelingualy patients treated with the same method.

Auditory Support in Linguistically Diverse Classrooms: Factors Related to Bilingual Text-to-Speech Use

ERIC Educational Resources Information Center

Van Laere, E.; Braak, J.

2017-01-01

Text-to-speech technology can act as an important support tool in computer-based learning environments (CBLEs) as it provides auditory input, next to on-screen text. Particularly for students who use a language at home other than the language of instruction (LOI) applied at school, text-to-speech can be useful. The CBLE E-Validiv offers content in…

Integrated Speech and Language Technology for Intelligence, Surveillance, and Reconnaissance (ISR)

DTIC Science & Technology

2017-07-01

applying submodularity techniques to address computing challenges posed by large datasets in speech and language processing. MT and speech tools were...aforementioned research-oriented activities, the IT system administration team provided necessary support to laboratory computing and network operations...operations of SCREAM Lab computer systems and networks. Other miscellaneous activities in relation to Task Order 29 are presented in an additional fourth

Intelligibility and Acceptability Testing for Speech Technology

DTIC Science & Technology

1992-05-22

information in memory (Luce, Feustel, and Pisoni, 1983). In high workload or multiple task situations, the added effort of listening to degraded speech can lead...the DRT provides diagnostic feature scores on six phonemic features: voicing, nasality, sustention , sibilation, graveness, and compactness, and on a...of other speech materials (e.g., polysyllabic words, paragraphs) and methods ( memory , comprehension, reaction time) have been used to evaluate the

Performance Evaluation of Intelligent Systems at the National Institute of Standards and Technology (NIST)

DTIC Science & Technology

2011-03-01

past few years, including performance evaluation of emergency response robots , sensor systems on unmanned ground vehicles, speech-to-speech translation...emergency response robots ; intelligent systems; mixed palletizing, testing, simulation; robotic vehicle perception systems; search and rescue robots ...ranging from autonomous vehicles to urban search and rescue robots to speech translation and manufacturing systems. The evaluations have occurred in

Nazareth College: Specialty Preparation for Speech-Language Pathologists to Work with Children Who Are Deaf and Hard of Hearing

ERIC Educational Resources Information Center

Brown, Paula M.; Quenin, Cathy

2010-01-01

The specialty preparation program within the speech-language pathology master's degree program at Nazareth College in Rochester, New York, was designed to train speech-language pathologists to work with children who are deaf and hard of hearing, ages 0 to 21. The program is offered in collaboration with the Rochester Institute of Technology,…

School Leavers with Learning Disabilities Moving from Child to Adult Speech and Language Therapy (SLT) Teams: SLTs' Views of Successful and Less Successful Transition Co-Working Practices

ERIC Educational Resources Information Center

McCartney, Elspeth; Muir, Margaret

2017-01-01

School-leaving for pupils with long-term speech, language, swallowing or communication difficulties requires careful management. Speech and language therapists (SLTs) support communication, secure assistive technology and manage swallowing difficulties post-school. UK SLTs are employed by health services, with child SLT teams based in schools.…

Optical chirp z-transform processor with a simplified architecture.

PubMed

Ngo, Nam Quoc

2014-12-29

Using a simplified chirp z-transform (CZT) algorithm based on the discrete-time convolution method, this paper presents the synthesis of a simplified architecture of a reconfigurable optical chirp z-transform (OCZT) processor based on the silica-based planar lightwave circuit (PLC) technology. In the simplified architecture of the reconfigurable OCZT, the required number of optical components is small and there are no waveguide crossings which make fabrication easy. The design of a novel type of optical discrete Fourier transform (ODFT) processor as a special case of the synthesized OCZT is then presented to demonstrate its effectiveness. The designed ODFT can be potentially used as an optical demultiplexer at the receiver of an optical fiber orthogonal frequency division multiplexing (OFDM) transmission system.

New developments for SAW channelization for mobile satellite payloads

NASA Technical Reports Server (NTRS)

Peach, R. C.; Mabson, P.

1995-01-01

The use of SAW technology in mobile communication payloads is becoming widely accepted by the industry since being pioneered by Inmarsat for its third generation of satellites. This paper presents new developments in this area, including broadband processors of the Inmarsat 3 type, and the use of SAW filters at L-band. It is demonstrated that SAW processors have considerable potential for increasing the capacity of future communications payloads, while allowing fully transparent operation without any restriction on traffic type or modulation format. In addition to the evolutionary development of Inmarsat type processors, new SAW applications have also emerged recently. Therefore, despite the rapid changes in the industry, it is predicted that SAW processing has a strong future in satellite communications.

A Series of Case Studies of Tinnitus Suppression With Mixed Background Stimuli in a Cochlear Implant

PubMed Central

Keiner, A. J.; Walker, Kurt; Deshpande, Aniruddha K.; Witt, Shelley; Killian, Matthijs; Ji, Helena; Patrick, Jim; Dillier, Norbert; van Dijk, Pim; Lai, Wai Kong; Hansen, Marlan R.; Gantz, Bruce

2015-01-01

Purpose Background sounds provided by a wearable sound playback device were mixed with the acoustical input picked up by a cochlear implant speech processor in an attempt to suppress tinnitus. Method First, patients were allowed to listen to several sounds and to select up to 4 sounds that they thought might be effective. These stimuli were programmed to loop continuously in the wearable playback device. Second, subjects were instructed to use 1 background sound each day on the wearable device, and they sequenced the selected background sounds during a 28-day trial. Patients were instructed to go to a website at the end of each day and rate the loudness and annoyance of the tinnitus as well as the acceptability of the background sound. Patients completed the Tinnitus Primary Function Questionnaire (Tyler, Stocking, Secor, & Slattery, 2014) at the beginning of the trial. Results Results indicated that background sounds were very effective at suppressing tinnitus. There was considerable variability in sounds preferred by the subjects. Conclusion The study shows that a background sound mixed with the microphone input can be effective for suppressing tinnitus during daily use of the sound processor in selected cochlear implant users. PMID:26001407

A practical method of predicting the loudness of complex electrical stimuli

NASA Astrophysics Data System (ADS)

McKay, Colette M.; Henshall, Katherine R.; Farrell, Rebecca J.; McDermott, Hugh J.

2003-04-01

The output of speech processors for multiple-electrode cochlear implants consists of current waveforms with complex temporal and spatial patterns. The majority of existing processors output sequential biphasic current pulses. This paper describes a practical method of calculating loudness estimates for such stimuli, in addition to the relative loudness contributions from different cochlear regions. The method can be used either to manipulate the loudness or levels in existing processing strategies, or to control intensity cues in novel sound processing strategies. The method is based on a loudness model described by McKay et al. [J. Acoust. Soc. Am. 110, 1514-1524 (2001)] with the addition of the simplifying approximation that current pulses falling within a temporal integration window of several milliseconds' duration contribute independently to the overall loudness of the stimulus. Three experiments were carried out with six implantees who use the CI24M device manufactured by Cochlear Ltd. The first experiment validated the simplifying assumption, and allowed loudness growth functions to be calculated for use in the loudness prediction method. The following experiments confirmed the accuracy of the method using multiple-electrode stimuli with various patterns of electrode locations and current levels.

High-Performance, Radiation-Hardened Electronics for Space Environments

NASA Technical Reports Server (NTRS)

Keys, Andrew S.; Watson, Michael D.; Frazier, Donald O.; Adams, James H.; Johnson, Michael A.; Kolawa, Elizabeth A.

2007-01-01

The Radiation Hardened Electronics for Space Environments (RHESE) project endeavors to advance the current state-of-the-art in high-performance, radiation-hardened electronics and processors, ensuring successful performance of space systems required to operate within extreme radiation and temperature environments. Because RHESE is a project within the Exploration Technology Development Program (ETDP), RHESE's primary customers will be the human and robotic missions being developed by NASA's Exploration Systems Mission Directorate (ESMD) in partial fulfillment of the Vision for Space Exploration. Benefits are also anticipated for NASA's science missions to planetary and deep-space destinations. As a technology development effort, RHESE provides a broad-scoped, full spectrum of approaches to environmentally harden space electronics, including new materials, advanced design processes, reconfigurable hardware techniques, and software modeling of the radiation environment. The RHESE sub-project tasks are: SelfReconfigurable Electronics for Extreme Environments, Radiation Effects Predictive Modeling, Radiation Hardened Memory, Single Event Effects (SEE) Immune Reconfigurable Field Programmable Gate Array (FPGA) (SIRF), Radiation Hardening by Software, Radiation Hardened High Performance Processors (HPP), Reconfigurable Computing, Low Temperature Tolerant MEMS by Design, and Silicon-Germanium (SiGe) Integrated Electronics for Extreme Environments. These nine sub-project tasks are managed by technical leads as located across five different NASA field centers, including Ames Research Center, Goddard Space Flight Center, the Jet Propulsion Laboratory, Langley Research Center, and Marshall Space Flight Center. The overall RHESE integrated project management responsibility resides with NASA's Marshall Space Flight Center (MSFC). Initial technology development emphasis within RHESE focuses on the hardening of Field Programmable Gate Arrays (FPGA)s and Field Programmable Analog Arrays (FPAA)s for use in reconfigurable architectures. As these component/chip level technologies mature, the RHESE project emphasis shifts to focus on efforts encompassing total processor hardening techniques and board-level electronic reconfiguration techniques featuring spare and interface modularity. This phased approach to distributing emphasis between technology developments provides hardened FPGA/FPAAs for early mission infusion, then migrates to hardened, board-level, high speed processors with associated memory elements and high density storage for the longer duration missions encountered for Lunar Outpost and Mars Exploration occurring later in the Constellation schedule.

Spacecraft computer technology at Southwest Research Institute

NASA Technical Reports Server (NTRS)

Shirley, D. J.

1993-01-01

Southwest Research Institute (SwRI) has developed and delivered spacecraft computers for a number of different near-Earth-orbit spacecraft including shuttle experiments and SDIO free-flyer experiments. We describe the evolution of the basic SwRI spacecraft computer design from those weighing in at 20 to 25 lb and using 20 to 30 W to newer models weighing less than 5 lb and using only about 5 W, yet delivering twice the processing throughput. Because of their reduced size, weight, and power, these newer designs are especially applicable to planetary instrument requirements. The basis of our design evolution has been the availability of more powerful processor chip sets and the development of higher density packaging technology, coupled with more aggressive design strategies in incorporating high-density FPGA technology and use of high-density memory chips. In addition to reductions in size, weight, and power, the newer designs also address the necessity of survival in the harsh radiation environment of space. Spurred by participation in such programs as MSTI, LACE, RME, Delta 181, Delta Star, and RADARSAT, our designs have evolved in response to program demands to be small, low-powered units, radiation tolerant enough to be suitable for both Earth-orbit microsats and for planetary instruments. Present designs already include MIL-STD-1750 and Multi-Chip Module (MCM) technology with near-term plans to include RISC processors and higher-density MCM's. Long term plans include development of whole-core processors on one or two MCM's.

Writing Centers in 2020--Gone!

ERIC Educational Resources Information Center

Bateman, Thomas L.

Technology brought the writing center to life because of the word processor, but new technology is actually going to create robotic life that thinks with us, for us, to us. It will offer portability all from a microchip stored in a coat pocket. Technology will continue to expedite today's hurry up world, and this will carry over into the writer's…

Prototyping the HPDP Chip on STM 65 NM Process

NASA Astrophysics Data System (ADS)

Papadas, C.; Dramitinos, G.; Syed, M.; Helfers, T.; Dedes, G.; Schoellkopf, J.-P.; Dugoujon, L.

2011-08-01

Currently Astrium GmbH is involved in the of the High Performance Data Processor (HPDP) development programme for telecommunication applications under a DLR contract. The HPDP project targets the implementation of the commercially available reconfigurable array processor IP (XPP from the company PACT XPP Technologies) in a radiation hardened technology.In the current complementary development phase funded under the Greek Industry Incentive scheme, it is planned to prototype the HPDP chip in commercial STM 65 nm technology. In addition it is also planned to utilise the preliminary radiation hardened components of this library wherever possible.This abstract gives an overview of the HPDP chip architecture, the basic details of the STM 65 nm process and the design flow foreseen for the prototyping. The paper will discuss the development and integration issues involved in using the STM 65 nm process (also including the available preliminary radiation hardened components) for designs targeted to be used in space applications.

Advanced digital SAR processing study

NASA Technical Reports Server (NTRS)

Martinson, L. W.; Gaffney, B. P.; Liu, B.; Perry, R. P.; Ruvin, A.

1982-01-01

A highly programmable, land based, real time synthetic aperture radar (SAR) processor requiring a processed pixel rate of 2.75 MHz or more in a four look system was designed. Variations in range and azimuth compression, number of looks, range swath, range migration and SR mode were specified. Alternative range and azimuth processing algorithms were examined in conjunction with projected integrated circuit, digital architecture, and software technologies. The advaced digital SAR processor (ADSP) employs an FFT convolver algorithm for both range and azimuth processing in a parallel architecture configuration. Algorithm performace comparisons, design system design, implementation tradeoffs and the results of a supporting survey of integrated circuit and digital architecture technologies are reported. Cost tradeoffs and projections with alternate implementation plans are presented.

Technology and design of an active-matrix OLED on crystalline silicon direct-view display for a wristwatch computer

NASA Astrophysics Data System (ADS)

Sanford, James L.; Schlig, Eugene S.; Prache, Olivier; Dove, Derek B.; Ali, Tariq A.; Howard, Webster E.

2002-02-01

The IBM Research Division and eMagin Corp. jointly have developed a low-power VGA direct view active matrix OLED display, fabricated on a crystalline silicon CMOS chip. The display is incorporated in IBM prototype wristwatch computers running the Linus operating system. IBM designed the silicon chip and eMagin developed the organic stack and performed the back-end-of line processing and packaging. Each pixel is driven by a constant current source controlled by a CMOS RAM cell, and the display receives its data from the processor memory bus. This paper describes the OLED technology and packaging, and outlines the design of the pixel and display electronics and the processor interface. Experimental results are presented.

Secondary Processors and Landfills — Partnerships that Work

NASA Astrophysics Data System (ADS)

Brewer, Ben; Roth, David J.

Using Best Available Technology is a phase that we often hear when there are environmental discussions on aluminum dross and secondary salt slag processing. The reality is best available technology is a mix between efficient removal of the valuable aluminum, oxides, misc metals and flux from dross and salt cake. This combined with conscientious land fill disposal of those items that finally, at this time, have no economic use is the reality of a company's best available actions. Recycling processes must be looked at with both the economic and environmental benefits weighed for their responsible implementation. This paper will discuss how this is done on a practical basis by Recycling Ventures (a secondary processor) and Environmental Waste Solutions (a Title II landfill), for the aluminum industry.

Applying the System Component and Operationally Relevant Evaluation (SCORE) Framework to Evaluate Advanced Military Technologies

DTIC Science & Technology

2010-03-01

and charac- terize the actions taken by the soldier (e.g., running, walking, climbing stairs ). Real-time image capture and exchange N The ability of...multimedia information sharing among soldiers in the field, two-way speech translation systems, and autonomous robotic platforms. Key words: Emerging...soldiers in the field, two-way speech translation systems, and autonomous robotic platforms. It has been the foundation for 10 technology evaluations

Using speech recognition to enhance the Tongue Drive System functionality in computer access.

PubMed

Huo, Xueliang; Ghovanloo, Maysam

2011-01-01

Tongue Drive System (TDS) is a wireless tongue operated assistive technology (AT), which can enable people with severe physical disabilities to access computers and drive powered wheelchairs using their volitional tongue movements. TDS offers six discrete commands, simultaneously available to the users, for pointing and typing as a substitute for mouse and keyboard in computer access, respectively. To enhance the TDS performance in typing, we have added a microphone, an audio codec, and a wireless audio link to its readily available 3-axial magnetic sensor array, and combined it with a commercially available speech recognition software, the Dragon Naturally Speaking, which is regarded as one of the most efficient ways for text entry. Our preliminary evaluations indicate that the combined TDS and speech recognition technologies can provide end users with significantly higher performance than using each technology alone, particularly in completing tasks that require both pointing and text entry, such as web surfing.

Missile signal processing common computer architecture for rapid technology upgrade

NASA Astrophysics Data System (ADS)

Rabinkin, Daniel V.; Rutledge, Edward; Monticciolo, Paul

2004-10-01

Interceptor missiles process IR images to locate an intended target and guide the interceptor towards it. Signal processing requirements have increased as the sensor bandwidth increases and interceptors operate against more sophisticated targets. A typical interceptor signal processing chain is comprised of two parts. Front-end video processing operates on all pixels of the image and performs such operations as non-uniformity correction (NUC), image stabilization, frame integration and detection. Back-end target processing, which tracks and classifies targets detected in the image, performs such algorithms as Kalman tracking, spectral feature extraction and target discrimination. In the past, video processing was implemented using ASIC components or FPGAs because computation requirements exceeded the throughput of general-purpose processors. Target processing was performed using hybrid architectures that included ASICs, DSPs and general-purpose processors. The resulting systems tended to be function-specific, and required custom software development. They were developed using non-integrated toolsets and test equipment was developed along with the processor platform. The lifespan of a system utilizing the signal processing platform often spans decades, while the specialized nature of processor hardware and software makes it difficult and costly to upgrade. As a result, the signal processing systems often run on outdated technology, algorithms are difficult to update, and system effectiveness is impaired by the inability to rapidly respond to new threats. A new design approach is made possible three developments; Moore's Law - driven improvement in computational throughput; a newly introduced vector computing capability in general purpose processors; and a modern set of open interface software standards. Today's multiprocessor commercial-off-the-shelf (COTS) platforms have sufficient throughput to support interceptor signal processing requirements. This application may be programmed under existing real-time operating systems using parallel processing software libraries, resulting in highly portable code that can be rapidly migrated to new platforms as processor technology evolves. Use of standardized development tools and 3rd party software upgrades are enabled as well as rapid upgrade of processing components as improved algorithms are developed. The resulting weapon system will have a superior processing capability over a custom approach at the time of deployment as a result of a shorter development cycles and use of newer technology. The signal processing computer may be upgraded over the lifecycle of the weapon system, and can migrate between weapon system variants enabled by modification simplicity. This paper presents a reference design using the new approach that utilizes an Altivec PowerPC parallel COTS platform. It uses a VxWorks-based real-time operating system (RTOS), and application code developed using an efficient parallel vector library (PVL). A quantification of computing requirements and demonstration of interceptor algorithm operating on this real-time platform are provided.

Evaluation of reinitialization-free nonvolatile computer systems for energy-harvesting Internet of things applications

NASA Astrophysics Data System (ADS)

Onizawa, Naoya; Tamakoshi, Akira; Hanyu, Takahiro

2017-08-01

In this paper, reinitialization-free nonvolatile computer systems are designed and evaluated for energy-harvesting Internet of things (IoT) applications. In energy-harvesting applications, as power supplies generated from renewable power sources cause frequent power failures, data processed need to be backed up when power failures occur. Unless data are safely backed up before power supplies diminish, reinitialization processes are required when power supplies are recovered, which results in low energy efficiencies and slow operations. Using nonvolatile devices in processors and memories can realize a faster backup than a conventional volatile computer system, leading to a higher energy efficiency. To evaluate the energy efficiency upon frequent power failures, typical computer systems including processors and memories are designed using 90 nm CMOS or CMOS/magnetic tunnel junction (MTJ) technologies. Nonvolatile ARM Cortex-M0 processors with 4 kB MRAMs are evaluated using a typical computing benchmark program, Dhrystone, which shows a few order-of-magnitude reductions in energy in comparison with a volatile processor with SRAM.

Multi-petascale highly efficient parallel supercomputer

DOEpatents

Asaad, Sameh; Bellofatto, Ralph E.; Blocksome, Michael A.; Blumrich, Matthias A.; Boyle, Peter; Brunheroto, Jose R.; Chen, Dong; Cher, Chen -Yong; Chiu, George L.; Christ, Norman; Coteus, Paul W.; Davis, Kristan D.; Dozsa, Gabor J.; Eichenberger, Alexandre E.; Eisley, Noel A.; Ellavsky, Matthew R.; Evans, Kahn C.; Fleischer, Bruce M.; Fox, Thomas W.; Gara, Alan; Giampapa, Mark E.; Gooding, Thomas M.; Gschwind, Michael K.; Gunnels, John A.; Hall, Shawn A.; Haring, Rudolf A.; Heidelberger, Philip; Inglett, Todd A.; Knudson, Brant L.; Kopcsay, Gerard V.; Kumar, Sameer; Mamidala, Amith R.; Marcella, James A.; Megerian, Mark G.; Miller, Douglas R.; Miller, Samuel J.; Muff, Adam J.; Mundy, Michael B.; O'Brien, John K.; O'Brien, Kathryn M.; Ohmacht, Martin; Parker, Jeffrey J.; Poole, Ruth J.; Ratterman, Joseph D.; Salapura, Valentina; Satterfield, David L.; Senger, Robert M.; Smith, Brian; Steinmacher-Burow, Burkhard; Stockdell, William M.; Stunkel, Craig B.; Sugavanam, Krishnan; Sugawara, Yutaka; Takken, Todd E.; Trager, Barry M.; Van Oosten, James L.; Wait, Charles D.; Walkup, Robert E.; Watson, Alfred T.; Wisniewski, Robert W.; Wu, Peng

2015-07-14

A Multi-Petascale Highly Efficient Parallel Supercomputer of 100 petaOPS-scale computing, at decreased cost, power and footprint, and that allows for a maximum packaging density of processing nodes from an interconnect point of view. The Supercomputer exploits technological advances in VLSI that enables a computing model where many processors can be integrated into a single Application Specific Integrated Circuit (ASIC). Each ASIC computing node comprises a system-on-chip ASIC utilizing four or more processors integrated into one die, with each having full access to all system resources and enabling adaptive partitioning of the processors to functions such as compute or messaging I/O on an application by application basis, and preferably, enable adaptive partitioning of functions in accordance with various algorithmic phases within an application, or if I/O or other processors are underutilized, then can participate in computation or communication nodes are interconnected by a five dimensional torus network with DMA that optimally maximize the throughput of packet communications between nodes and minimize latency.

Resource and Performance Evaluations of Fixed Point QRD-RLS Systolic Array through FPGA Implementation

NASA Astrophysics Data System (ADS)

Yokoyama, Yoshiaki; Kim, Minseok; Arai, Hiroyuki

At present, when using space-time processing techniques with multiple antennas for mobile radio communication, real-time weight adaptation is necessary. Due to the progress of integrated circuit technology, dedicated processor implementation with ASIC or FPGA can be employed to implement various wireless applications. This paper presents a resource and performance evaluation of the QRD-RLS systolic array processor based on fixed-point CORDIC algorithm with FPGA. In this paper, to save hardware resources, we propose the shared architecture of a complex CORDIC processor. The required precision of internal calculation, the circuit area for the number of antenna elements and wordlength, and the processing speed will be evaluated. The resource estimation provides a possible processor configuration with a current FPGA on the market. Computer simulations assuming a fading channel will show a fast convergence property with a finite number of training symbols. The proposed architecture has also been implemented and its operation was verified by beamforming evaluation through a radio propagation experiment.

Discover Supercomputer 5

NASA Image and Video Library

2017-12-08

Two rows of the “Discover” supercomputer at the NASA Center for Climate Simulation (NCCS) contain more than 4,000 computer processors. Discover has a total of nearly 15,000 processors. Credit: NASA/Pat Izzo To learn more about NCCS go to: www.nasa.gov/topics/earth/features/climate-sim-center.html NASA Goddard Space Flight Center is home to the nation's largest organization of combined scientists, engineers and technologists that build spacecraft, instruments and new technology to study the Earth, the sun, our solar system, and the universe.

120-MHz BiCMOS superscalar RISC processor

NASA Astrophysics Data System (ADS)

Tanaka, Shigeya; Hotta, Takashi; Murabayashi, Fumio; Yamada, Hiromichi; Yoshida, Shoji; Shimamura, Kotaro; Katsura, Koyo; Bandoh, Tadaaki; Ikeda, Koichi; Matsubara, Kenji

1994-04-01

A superscalar RISC processor contains 2.8 million transistors in a die size of 16.2 mm x 16.5 mm, and utilizes 3.3 V/0.5 micron BiCMOS technology. In order to take advantage of superscalar performance without incurring penalties from a slower clock or a longer pipeline, a tag bit is implemented in the instruction cache to indicate dependency between two instructions. A performance gain of up to 37% is obtained with only a 3.5% area overhead from our superscalar design.

Using Technology to Learn from Travelmates' Adventures.

ERIC Educational Resources Information Center

Braun, Joseph A., Jr.; Kraft, Christine

1995-01-01

Describes a thematic curriculum unit in which elementary students simulate travel around the world and record their experiences using word processors and databases. Includes figures listing student directions and an actual student journal entry. Asserts that technology can heighten student motivation and improve knowledge about the world. (CFR)

The ISS Water Processor Catalytic Reactor as a Post Processor for Advanced Water Reclamation Systems

NASA Technical Reports Server (NTRS)

Nalette, Tim; Snowdon, Doug; Pickering, Karen D.; Callahan, Michael

2007-01-01

Advanced water processors being developed for NASA s Exploration Initiative rely on phase change technologies and/or biological processes as the primary means of water reclamation. As a result of the phase change, volatile compounds will also be transported into the distillate product stream. The catalytic reactor assembly used in the International Space Station (ISS) water processor assembly, referred to as Volatile Removal Assembly (VRA), has demonstrated high efficiency oxidation of many of these volatile contaminants, such as low molecular weight alcohols and acetic acid, and is considered a viable post treatment system for all advanced water processors. To support this investigation, two ersatz solutions were defined to be used for further evaluation of the VRA. The first solution was developed as part of an internal research and development project at Hamilton Sundstrand (HS) and is based primarily on ISS experience related to the development of the VRA. The second ersatz solution was defined by NASA in support of a study contract to Hamilton Sundstrand to evaluate the VRA as a potential post processor for the Cascade Distillation system being developed by Honeywell. This second ersatz solution contains several low molecular weight alcohols, organic acids, and several inorganic species. A range of residence times, oxygen concentrations and operating temperatures have been studied with both ersatz solutions to provide addition performance capability of the VRA catalyst.

Do What I Say! Voice Recognition Makes Major Advances.

ERIC Educational Resources Information Center

Ruley, C. Dorsey

1994-01-01

Explains voice recognition technology applications in the workplace, schools, and libraries. Highlights include a voice-controlled work station using the DragonDictate system that can be used with dyslexic students, converting text to speech, and converting speech to text. (LRW)

Voice Response Systems Technology.

ERIC Educational Resources Information Center

Gerald, Jeanette

1984-01-01

Examines two methods of generating synthetic speech in voice response systems, which allow computers to communicate in human terms (speech), using human interface devices (ears): phoneme and reconstructed voice systems. Considerations prior to implementation, current and potential applications, glossary, directory, and introduction to Input Output…

Advancements in text-to-speech technology and implications for AAC applications

NASA Astrophysics Data System (ADS)

Syrdal, Ann K.

2003-10-01

Intelligibility was the initial focus in text-to-speech (TTS) research, since it is clearly a necessary condition for the application of the technology. Sufficiently high intelligibility (approximating human speech) has been achieved in the last decade by the better formant-based and concatenative TTS systems. This led to commercially available TTS systems for highly motivated users, particularly the blind and vocally impaired. Some unnatural qualities of TTS were exploited by these users, such as very fast speaking rates and altered pitch ranges for flagging relevant information. Recently, the focus in TTS research has turned to improving naturalness, so that synthetic speech sounds more human and less robotic. Unit selection approaches to concatenative synthesis have dramatically improved TTS quality, although at the cost of larger and more complex systems. This advancement in naturalness has made TTS technology more acceptable to the general public. The vocally impaired appreciate a more natural voice with which to represent themselves when communicating with others. Unit selection TTS does not achieve such high speaking rates as the earlier TTS systems, however, which is a disadvantage to some AAC device users. An important new research emphasis is to improve and increase the range of emotional expressiveness of TTS.

DARPA TIMIT acoustic-phonetic continous speech corpus CD-ROM. NIST speech disc 1-1.1

NASA Astrophysics Data System (ADS)

Garofolo, J. S.; Lamel, L. F.; Fisher, W. M.; Fiscus, J. G.; Pallett, D. S.

1993-02-01

The Texas Instruments/Massachusetts Institute of Technology (TIMIT) corpus of read speech has been designed to provide speech data for the acquisition of acoustic-phonetic knowledge and for the development and evaluation of automatic speech recognition systems. TIMIT contains speech from 630 speakers representing 8 major dialect divisions of American English, each speaking 10 phonetically-rich sentences. The TIMIT corpus includes time-aligned orthographic, phonetic, and word transcriptions, as well as speech waveform data for each spoken sentence. The release of TIMIT contains several improvements over the Prototype CD-ROM released in December, 1988: (1) full 630-speaker corpus, (2) checked and corrected transcriptions, (3) word-alignment transcriptions, (4) NIST SPHERE-headered waveform files and header manipulation software, (5) phonemic dictionary, (6) new test and training subsets balanced for dialectal and phonetic coverage, and (7) more extensive documentation.

Implications of Multilingual Interoperability of Speech Technology for Military Use (Les implications de l’interoperabilite multilingue des technologies vocales pour applications militaires)

DTIC Science & Technology

2004-09-01

Databases 2-2 2.3.1 Translanguage English Database 2-2 2.3.2 Australian National Database of Spoken Language 2-3 2.3.3 Strange Corpus 2-3 2.3.4...some relevance to speech technology research. 2.3.1 Translanguage English Database In a daring plan Joseph Mariani, then at LIMSI-CNRS, proposed to...native speakers. The database is known as the ‘ Translanguage English Database’ but is often referred to as the ‘terrible English database.’ About 28

MULTI-CORE AND OPTICAL PROCESSOR RELATED APPLICATIONS RESEARCH AT OAK RIDGE NATIONAL LABORATORY

DOE Office of Scientific and Technical Information (OSTI.GOV)

Barhen, Jacob; Kerekes, Ryan A; ST Charles, Jesse Lee

2008-01-01

High-speed parallelization of common tasks holds great promise as a low-risk approach to achieving the significant increases in signal processing and computational performance required for next generation innovations in reconfigurable radio systems. Researchers at the Oak Ridge National Laboratory have been working on exploiting the parallelization offered by this emerging technology and applying it to a variety of problems. This paper will highlight recent experience with four different parallel processors applied to signal processing tasks that are directly relevant to signal processing required for SDR/CR waveforms. The first is the EnLight Optical Core Processor applied to matched filter (MF) correlationmore » processing via fast Fourier transform (FFT) of broadband Dopplersensitive waveforms (DSW) using active sonar arrays for target tracking. The second is the IBM CELL Broadband Engine applied to 2-D discrete Fourier transform (DFT) kernel for image processing and frequency domain processing. And the third is the NVIDIA graphical processor applied to document feature clustering. EnLight Optical Core Processor. Optical processing is inherently capable of high-parallelism that can be translated to very high performance, low power dissipation computing. The EnLight 256 is a small form factor signal processing chip (5x5 cm2) with a digital optical core that is being developed by an Israeli startup company. As part of its evaluation of foreign technology, ORNL's Center for Engineering Science Advanced Research (CESAR) had access to a precursor EnLight 64 Alpha hardware for a preliminary assessment of capabilities in terms of large Fourier transforms for matched filter banks and on applications related to Doppler-sensitive waveforms. This processor is optimized for array operations, which it performs in fixed-point arithmetic at the rate of 16 TeraOPS at 8-bit precision. This is approximately 1000 times faster than the fastest DSP available today. The optical core performs the matrix-vector multiplications, where the nominal matrix size is 256x256. The system clock is 125MHz. At each clock cycle, 128K multiply-and-add operations per second (OPS) are carried out, which yields a peak performance of 16 TeraOPS. IBM Cell Broadband Engine. The Cell processor is the extraordinary resulting product of 5 years of sustained, intensive R&D collaboration (involving over $400M investment) between IBM, Sony, and Toshiba. Its architecture comprises one multithreaded 64-bit PowerPC processor element (PPE) with VMX capabilities and two levels of globally coherent cache, and 8 synergistic processor elements (SPEs). Each SPE consists of a processor (SPU) designed for streaming workloads, local memory, and a globally coherent direct memory access (DMA) engine. Computations are performed in 128-bit wide single instruction multiple data streams (SIMD). An integrated high-bandwidth element interconnect bus (EIB) connects the nine processors and their ports to external memory and to system I/O. The Applied Software Engineering Research (ASER) Group at the ORNL is applying the Cell to a variety of text and image analysis applications. Research on Cell-equipped PlayStation3 (PS3) consoles has led to the development of a correlation-based image recognition engine that enables a single PS3 to process images at more than 10X the speed of state-of-the-art single-core processors. NVIDIA Graphics Processing Units. The ASER group is also employing the latest NVIDIA graphical processing units (GPUs) to accelerate clustering of thousands of text documents using recently developed clustering algorithms such as document flocking and affinity propagation.« less

Machine Vision Giving Eyes to Robots. Resources in Technology.

ERIC Educational Resources Information Center

Technology Teacher, 1990

1990-01-01

This module introduces machine vision, which can be used for inspection, robot guidance and part sorting. The future for machine vision will include new technology and will bring vision systems closer to the ultimate vision processor, the human eye. Includes a student quiz, outcomes, and activities. (JOW)

Real-time interactive speech technology at Threshold Technology, Incorporated

NASA Technical Reports Server (NTRS)

Herscher, Marvin B.

1977-01-01

Basic real-time isolated-word recognition techniques are reviewed. Industrial applications of voice technology are described in chronological order of their development. Future research efforts are also discussed.

Automatic translation among spoken languages

NASA Technical Reports Server (NTRS)

Walter, Sharon M.; Costigan, Kelly

1994-01-01

The Machine Aided Voice Translation (MAVT) system was developed in response to the shortage of experienced military field interrogators with both foreign language proficiency and interrogation skills. Combining speech recognition, machine translation, and speech generation technologies, the MAVT accepts an interrogator's spoken English question and translates it into spoken Spanish. The spoken Spanish response of the potential informant can then be translated into spoken English. Potential military and civilian applications for automatic spoken language translation technology are discussed in this paper.

Improving medical imaging report turnaround times: the role of technolgy.

PubMed

Marquez, Luis O; Stewart, Howard

2005-01-01

At Southern Ohio Medical Center (SOMC), the medical imaging department and the radiologists expressed a strong desire to improve workflow. The improved workflow was a major motivating factor toward implementing a new RIS and speech recognition technology. The need to monitor workflow in a real-time fashion and to evaluate productivity and resources necessitated that a new solution be found. A decision was made to roll out both the new RIS product and speech recognition to maximize the resources to interface and implement the new solution. Prior to implementation of the new RIS, the medical imaging department operated in a conventional electronic-order-entry to paper request manner. The paper request followed the study through exam completion to the radiologist. SOMC entered into a contract with its PACS vendor to participate in beta testing and clinical trials for a new RIS product for the US market. Backup plans were created in the event the product failed to function as planned--either during the beta testing period or during clinical trails. The last piece of the technology puzzle to improve report turnaround time was voice recognition technology. Speech recognition enhanced the RIS technology as soon as it was implemented. The results show that the project has been a success. The new RIS, combined with speech recognition and the PACS, makes for a very effective solution to patient, exam, and results management in the medical imaging department.

Speaking with a mirror: engagement of mirror neurons via choral speech and its derivatives induces stuttering inhibition.

PubMed

Kalinowski, Joseph; Saltuklaroglu, Tim

2003-04-01

'Choral speech', 'unison speech', or 'imitation speech' has long been known to immediately induce reflexive, spontaneous, and natural sounding fluency, even the most severe cases of stuttering. Unlike typical post-therapeutic speech, a hallmark characteristic of choral speech is the sense of 'invulnerability' to stuttering, regardless of phonetic context, situational environment, or audience size. We suggest that choral speech immediately inhibits stuttering by engaging mirror systems of neurons, innate primitive neuronal substrates that dominate the initial phases of language development due to their predisposition to reflexively imitate gestural action sequences in a fluent manner. Since mirror systems are primordial in nature, they take precedence over the much later developing stuttering pathology. We suggest that stuttering may best be ameliorated by reengaging mirror neurons via choral speech or one of its derivatives (using digital signal processing technology) to provide gestural mirrors, that are nature's way of immediately overriding the central stuttering block. Copyright 2003 Elsevier Science Ltd.

On the recognition of emotional vocal expressions: motivations for a holistic approach.

PubMed

Esposito, Anna; Esposito, Antonietta M

2012-10-01

Human beings seem to be able to recognize emotions from speech very well and information communication technology aims to implement machines and agents that can do the same. However, to be able to automatically recognize affective states from speech signals, it is necessary to solve two main technological problems. The former concerns the identification of effective and efficient processing algorithms capable of capturing emotional acoustic features from speech sentences. The latter focuses on finding computational models able to classify, with an approximation as good as human listeners, a given set of emotional states. This paper will survey these topics and provide some insights for a holistic approach to the automatic analysis, recognition and synthesis of affective states.

Selecting cockpit functions for speech I/O technology

NASA Technical Reports Server (NTRS)

Simpson, C. A.

1985-01-01

A general methodology for the initial selection of functions for speech generation and speech recognition technology is discussed. The SCR (Stimulus/Central-Processing/Response) compatibility model of Wickens et al. (1983) is examined, and its application is demonstrated for a particular cockpit display problem. Some limits of the applicability of that model are illustrated in the context of predicting overall pilot-aircraft system performance. A program of system performance measurement is recommended for the evaluation of candidate systems. It is suggested that no one measure of system performance can necessarily be depended upon to the exclusion of others. Systems response time, system accuracy, and pilot ratings are all important measures. Finally, these measures must be collected in the context of the total flight task environment.

Research on NC motion controller based on SOPC technology

NASA Astrophysics Data System (ADS)

Jiang, Tingbiao; Meng, Biao

2006-11-01

With the rapid development of the digitization and informationization, the application of numerical control technology in the manufacturing industry becomes more and more important. However, the conventional numerical control system usually has some shortcomings such as the poor in system openness, character of real-time, cutability and reconfiguration. In order to solve these problems, this paper investigates the development prospect and advantage of the application in numerical control area with system-on-a-Programmable-Chip (SOPC) technology, and puts forward to a research program approach to the NC controller based on SOPC technology. Utilizing the characteristic of SOPC technology, we integrate high density logic device FPGA, memory SRAM, and embedded processor ARM into a single programmable logic device. We also combine the 32-bit RISC processor with high computing capability of the complicated algorithm with the FPGA device with strong motivable reconfiguration logic control ability. With these steps, we can greatly resolve the defect described in above existing numerical control systems. For the concrete implementation method, we use FPGA chip embedded with ARM hard nuclear processor to construct the control core of the motion controller. We also design the peripheral circuit of the controller according to the requirements of actual control functions, transplant real-time operating system into ARM, design the driver of the peripheral assisted chip, develop the application program to control and configuration of FPGA, design IP core of logic algorithm for various NC motion control to configured it into FPGA. The whole control system uses the concept of modular and structured design to develop hardware and software system. Thus the NC motion controller with the advantage of easily tailoring, highly opening, reconfigurable, and expandable can be implemented.

A High-Throughput Processor for Flight Control Research Using Small UAVs

NASA Technical Reports Server (NTRS)

Klenke, Robert H.; Sleeman, W. C., IV; Motter, Mark A.

2006-01-01

There are numerous autopilot systems that are commercially available for small (<100 lbs) UAVs. However, they all share several key disadvantages for conducting aerodynamic research, chief amongst which is the fact that most utilize older, slower, 8- or 16-bit microcontroller technologies. This paper describes the development and testing of a flight control system (FCS) for small UAV s based on a modern, high throughput, embedded processor. In addition, this FCS platform contains user-configurable hardware resources in the form of a Field Programmable Gate Array (FPGA) that can be used to implement custom, application-specific hardware. This hardware can be used to off-load routine tasks such as sensor data collection, from the FCS processor thereby further increasing the computational throughput of the system.

RISC Processors and High Performance Computing

NASA Technical Reports Server (NTRS)

Bailey, David H.; Saini, Subhash; Craw, James M. (Technical Monitor)

1995-01-01

This tutorial will discuss the top five RISC microprocessors and the parallel systems in which they are used. It will provide a unique cross-machine comparison not available elsewhere. The effective performance of these processors will be compared by citing standard benchmarks in the context of real applications. The latest NAS Parallel Benchmarks, both absolute performance and performance per dollar, will be listed. The next generation of the NPB will be described. The tutorial will conclude with a discussion of future directions in the field. Technology Transfer Considerations: All of these computer systems are commercially available internationally. Information about these processors is available in the public domain, mostly from the vendors themselves. The NAS Parallel Benchmarks and their results have been previously approved numerous times for public release, beginning back in 1991.

Companion Chip: Building a Segregated Hardware Architecture

NASA Astrophysics Data System (ADS)

Pareaud, Thomas; Houelle, Alain; Vaucher, Niolas; Albinet, Mathieu; Honvault, Christophe

2011-08-01

Partitioning is a more and more mature concept in Space industry. It aims at assuring that some error propagation modes are not possible. This paper gives an overview of an analysis conducted in the frame of a research and technology study performed in 2010/2011. The "Java Companion Chip" study addresses an interesting approach to partitioning using hardware concepts: a SoC architecture integrates a master processor, a companion chip and additional hardware functions aiming at enforcing the time and space segregation between the master processor and the slave one.This paper discusses the benefits and the main challenges of the proposed approach. In addition, it presents an application of these concepts to a case study: a Leon/Java processor architecture able to concurrently execute native and Java applications.

Light-Driven Contact Hearing Aid for Broad-Spectrum Amplification: Safety and Effectiveness Pivotal Study.

PubMed

Gantz, Bruce J; Perkins, Rodney; Murray, Michael; Levy, Suzanne Carr; Puria, Sunil

2017-03-01

Demonstrate safety and effectiveness of the light-driven contact hearing aid to support FDA clearance. A single-arm, open-label investigational-device clinical trial. Two private-practice and one hospital-based ENT clinics. Forty-three subjects (86 ears) with mild-to-severe bilateral sensorineural hearing impairment. Bilateral amplification delivered via a light-driven contact hearing aid comprising a Tympanic Lens (Lens) with a customized platform to directly drive the umbo and a behind-the-ear sound processor (Processor) that encodes sound into light pulses to wirelessly deliver signal and power to the Lens. The primary safety endpoint was a determination of "no change" (PTA4 < 10 dB) in residual unaided hearing at the 120-day measurement interval. The primary efficacy endpoint was improvement in word recognition using NU-6 at the 30-day measurement interval over the baseline unaided case. Secondary efficacy endpoints included functional gain from 2 to 10 kHz and speech-in-noise improvement over the baseline unaided case using both omnidirectional and directional microphones. The results for the 86 ears in the study determined a mean change of -0.40 dB in PTA4, indicating no change in residual hearing (p < 0.0001). There were no serious device- or procedure-related adverse events, or unanticipated adverse events. Word recognition aided with the Earlens improved significantly (p < 0.0001) over the unaided performance, by 35% rationalized arcsine units on average. Mean functional gain was 31 dB across 2 to 10 kHz. The average speech-recognition threshold improvement over the unaided case for the Hearing in Noise Test was 0.75 dB (p = 0.028) and 3.14 dB (p < 0.0001) for the omnidirectional and directional microphone modes, respectively. The safety and effectiveness data supported a de novo 510(k) submission that received clearance from the FDA.

Tolerable hearing-aid delays: IV. effects on subjective disturbance during speech production by hearing-impaired subjects.

PubMed

Stone, Michael A; Moore, Brian C J

2005-04-01

We assessed the effects of time delay in a hearing aid on subjective disturbance and reading rates while the user of the aid was speaking, using hearing-impaired subjects and real-time processing. The time delay was constant across frequency. A digital signal processor was programmed as a four-channel, fast-acting, wide-dynamic-range compression hearing aid. One of four delays could be selected on the aid to produce a total delay of 13, 21, 30, or 40 msec between microphone and receiver. Twenty-five subjects, mostly with near-symmetric hearing impairment of cochlear origin, were fitted bilaterally with behind-the-ear aids connected to the processor. The aids were programmed with insertion gains prescribed by the CAMEQ loudness equalization procedure for each subject and ear. Subjects were asked to read aloud from scripts: speech production rates were measured and subjective ratings of the disturbance of the delay were obtained. Subjects required some training to recognize the effects of the delay to rate it consistently. Subjective disturbance increased progressively with increasing delay and was a nonmonotonic function of low-frequency hearing loss. Subjects with mild or severe low-frequency hearing loss were generally less disturbed by the delay than those with moderate loss. Disturbance ratings tended to decrease over successive tests. Word production rates were not significantly affected by delay over the range of delays tested. The results follow a pattern similar to those presented in , obtained using a simulation of hearing loss and normally hearing subjects, except for the nonmonotonic variation of disturbance with low-frequency hearing loss. We hypothesize that disturbance is maximal when the levels in the ear canal of the low-frequency components are similar for the unaided and aided sounds. A rating of 3, which is probably just acceptable, was obtained for delays ranging from 14 to 30 msec, depending on the hearing loss. Some acclimatization to the subjective disturbance occurred over a time scale of about 1 hour.

Hands-free device control using sound picked up in the ear canal

NASA Astrophysics Data System (ADS)

Chhatpar, Siddharth R.; Ngia, Lester; Vlach, Chris; Lin, Dong; Birkhimer, Craig; Juneja, Amit; Pruthi, Tarun; Hoffman, Orin; Lewis, Tristan

2008-04-01

Hands-free control of unmanned ground vehicles is essential for soldiers, bomb disposal squads, and first responders. Having their hands free for other equipment and tasks allows them to be safer and more mobile. Currently, the most successful hands-free control devices are speech-command based. However, these devices use external microphones, and in field environments, e.g., war zones and fire sites, their performance suffers because of loud ambient noise: typically above 90dBA. This paper describes the development of technology using the ear as an output source that can provide excellent command recognition accuracy even in noisy environments. Instead of picking up speech radiating from the mouth, this technology detects speech transmitted internally through the ear canal. Discreet tongue movements also create air pressure changes within the ear canal, and can be used for stealth control. A patented earpiece was developed with a microphone pointed into the ear canal that captures these signals generated by tongue movements and speech. The signals are transmitted from the earpiece to an Ultra-Mobile Personal Computer (UMPC) through a wired connection. The UMPC processes the signals and utilizes them for device control. The processing can include command recognition, ambient noise cancellation, acoustic echo cancellation, and speech equalization. Successful control of an iRobot PackBot has been demonstrated with both speech (13 discrete commands) and tongue (5 discrete commands) signals. In preliminary tests, command recognition accuracy was 95% with speech control and 85% with tongue control.

Feasibility of optically interconnected parallel processors using wavelength division multiplexing

DOE Office of Scientific and Technical Information (OSTI.GOV)

Deri, R.J.; De Groot, A.J.; Haigh, R.E.

1996-03-01

New national security demands require enhanced computing systems for nearly ab initio simulations of extremely complex systems and analyzing unprecedented quantities of remote sensing data. This computational performance is being sought using parallel processing systems, in which many less powerful processors are ganged together to achieve high aggregate performance. Such systems require increased capability to communicate information between individual processor and memory elements. As it is likely that the limited performance of today`s electronic interconnects will prevent the system from achieving its ultimate performance, there is great interest in using fiber optic technology to improve interconnect communication. However, little informationmore » is available to quantify the requirements on fiber optical hardware technology for this application. Furthermore, we have sought to explore interconnect architectures that use the complete communication richness of the optical domain rather than using optics as a simple replacement for electronic interconnects. These considerations have led us to study the performance of a moderate size parallel processor with optical interconnects using multiple optical wavelengths. We quantify the bandwidth, latency, and concurrency requirements which allow a bus-type interconnect to achieve scalable computing performance using up to 256 nodes, each operating at GFLOP performance. Our key conclusion is that scalable performance, to {approx}150 GFLOPS, is achievable for several scientific codes using an optical bus with a small number of WDM channels (8 to 32), only one WDM channel received per node, and achievable optoelectronic bandwidth and latency requirements. 21 refs. , 10 figs.« less

A Biologically-Based Alternative Water Processor for Long Duration Space Missions

NASA Technical Reports Server (NTRS)

Barta, Daniel J.; Pickering, Karen D.; Meyer, Caitlin; Pensinger, Stuart; Vega, Leticia; Flynn, Michael; Jackson, Andrew; Wheeler, Raymond

2015-01-01

A wastewater recovery system has been developed that combines novel biological and physicochemical components for recycling wastewater on long duration space missions. Functionally, this Alternative Water Processor (AWP) would replace the Urine Processing Assembly on the International Space Station and reduce or eliminate the need for the multifiltration beds of the Water Processing Assembly (WPA). At its center are two unique game changing technologies: 1) a biological water processor (BWP) to mineralize organic forms of carbon and nitrogen and 2) an advanced membrane processor (Forward Osmosis Secondary Treatment) for removal of solids and inorganic ions. The AWP is designed for recycling larger quantities of wastewater from multiple sources expected during future exploration missions, including urine, hygiene (hand wash, shower, oral and shave) and laundry. The BWP utilizes a single-stage membrane-aerated biological reactor for simultaneous nitrification and denitrification. The Forward Osmosis Secondary Treatment (FOST) system uses a combination of forward osmosis (FO) and reverse osmosis (RO), is resistant to biofouling and can easily tolerate wastewaters high in non-volatile organics and solids associated with shower and/or hand washing. The BWP was operated continuously for over 300 days. After startup, the mature biological system averaged 85% organic carbon removal and 44% nitrogen removal, close to maximum based on available carbon. The FOST has averaged 93% water recovery, with a maximum of 98%. If the wastewater is slighty acidified, ammonia rejection is optimal. This paper will provide a description of the technology and summarize results from ground-based testing using real wastewater.

Hearing aid and hearing assistance technology use in Aotearoa/New Zealand.

PubMed

Kelly-Campbell, Rebecca J; Lessoway, Kamea

2015-05-01

The purpose of this study was to describe factors that are related to hearing aid and hearing assistance technology ownership and use in Aotearoa/New Zealand. Adults with hearing impairment living in New Zealand were surveyed regarding health-related quality of life and device usage. Audiometric data (hearing sensitivity and speech in noise) were collected. Data were obtained from 123 adults with hearing impairment: 73 reported current hearing-aid use, 81 reported current hearing assistance technology use. In both analyses, device users had more difficulty understanding speech in background noise, had poor hearing in both their better and worse hearing ears, and perceived more consequences of hearing impairment in their everyday lives (both emotionally and socially) than non-hearing-aid users. Discriminant analyses showed that the social consequences of hearing impairment and the better ear hearing best classified hearing aid users from non-users but social consequences and worse ear hearing best classified hearing assistance technology users from non-users. Quality of life measurements and speech-in-noise assessments provide useful clinical information. Hearing-impaired adults in New Zealand who use hearing aids also tend to use hearing assistance technology, which has important clinical implications.

Speech and gesture interfaces for squad-level human-robot teaming

NASA Astrophysics Data System (ADS)

Harris, Jonathan; Barber, Daniel

2014-06-01

As the military increasingly adopts semi-autonomous unmanned systems for military operations, utilizing redundant and intuitive interfaces for communication between Soldiers and robots is vital to mission success. Currently, Soldiers use a common lexicon to verbally and visually communicate maneuvers between teammates. In order for robots to be seamlessly integrated within mixed-initiative teams, they must be able to understand this lexicon. Recent innovations in gaming platforms have led to advancements in speech and gesture recognition technologies, but the reliability of these technologies for enabling communication in human robot teaming is unclear. The purpose for the present study is to investigate the performance of Commercial-Off-The-Shelf (COTS) speech and gesture recognition tools in classifying a Squad Level Vocabulary (SLV) for a spatial navigation reconnaissance and surveillance task. The SLV for this study was based on findings from a survey conducted with Soldiers at Fort Benning, GA. The items of the survey focused on the communication between the Soldier and the robot, specifically in regards to verbally instructing them to execute reconnaissance and surveillance tasks. Resulting commands, identified from the survey, were then converted to equivalent arm and hand gestures, leveraging existing visual signals (e.g. U.S. Army Field Manual for Visual Signaling). A study was then run to test the ability of commercially available automated speech recognition technologies and a gesture recognition glove to classify these commands in a simulated intelligence, surveillance, and reconnaissance task. This paper presents classification accuracy of these devices for both speech and gesture modalities independently.

Predicting Cost/Performance Trade-Offs for Whitney: A Commodity Computing Cluster

NASA Technical Reports Server (NTRS)

Becker, Jeffrey C.; Nitzberg, Bill; VanderWijngaart, Rob F.; Kutler, Paul (Technical Monitor)

1997-01-01

Recent advances in low-end processor and network technology have made it possible to build a "supercomputer" out of commodity components. We develop simple models of the NAS Parallel Benchmarks version 2 (NPB 2) to explore the cost/performance trade-offs involved in building a balanced parallel computer supporting a scientific workload. We develop closed form expressions detailing the number and size of messages sent by each benchmark. Coupling these with measured single processor performance, network latency, and network bandwidth, our models predict benchmark performance to within 30%. A comparison based on total system cost reveals that current commodity technology (200 MHz Pentium Pros with 100baseT Ethernet) is well balanced for the NPBs up to a total system cost of around $1,000,000.

A Comparative Analysis of Phase-Change Wastewater Processing Approaches for Microgravity

NASA Technical Reports Server (NTRS)

Lange, Kevin

2016-01-01

Two phase-change wastewater processing candidates, the ISS Vapor Compression Distillation (VCD) System and the Cascade Distiller System (CDS), are compared based on dynamic modeling of both technologies. Differences in fluid handling and energy recovery for the technologies are described and contrasted. Model predictions are presented showing how temperatures, pressures, and compositions vary locally within each distiller. These dynamic variations are difficult to observe experimentally and have implications regarding non-condensable buildup and salt precipitation potential. Alternative architectures involving VCD and CDS components are analyzed in terms of predicted performance and equivalent system mass (ESM). The addition of a downstream brine processor to increase water recovery is also evaluated. Options for reducing overall ESM are discussed, including the possibility of developing a single precipitation-tolerant primary wastewater processor.

Strategies for distant speech recognitionin reverberant environments

NASA Astrophysics Data System (ADS)

Delcroix, Marc; Yoshioka, Takuya; Ogawa, Atsunori; Kubo, Yotaro; Fujimoto, Masakiyo; Ito, Nobutaka; Kinoshita, Keisuke; Espi, Miquel; Araki, Shoko; Hori, Takaaki; Nakatani, Tomohiro

2015-12-01

Reverberation and noise are known to severely affect the automatic speech recognition (ASR) performance of speech recorded by distant microphones. Therefore, we must deal with reverberation if we are to realize high-performance hands-free speech recognition. In this paper, we review a recognition system that we developed at our laboratory to deal with reverberant speech. The system consists of a speech enhancement (SE) front-end that employs long-term linear prediction-based dereverberation followed by noise reduction. We combine our SE front-end with an ASR back-end that uses neural networks for acoustic and language modeling. The proposed system achieved top scores on the ASR task of the REVERB challenge. This paper describes the different technologies used in our system and presents detailed experimental results that justify our implementation choices and may provide hints for designing distant ASR systems.

Intelligent systems technology infrastructure for integrated systems

NASA Technical Reports Server (NTRS)

Lum, Henry, Jr.

1991-01-01

Significant advances have occurred during the last decade in intelligent systems technologies (a.k.a. knowledge-based systems, KBS) including research, feasibility demonstrations, and technology implementations in operational environments. Evaluation and simulation data obtained to date in real-time operational environments suggest that cost-effective utilization of intelligent systems technologies can be realized for Automated Rendezvous and Capture applications. The successful implementation of these technologies involve a complex system infrastructure integrating the requirements of transportation, vehicle checkout and health management, and communication systems without compromise to systems reliability and performance. The resources that must be invoked to accomplish these tasks include remote ground operations and control, built-in system fault management and control, and intelligent robotics. To ensure long-term evolution and integration of new validated technologies over the lifetime of the vehicle, system interfaces must also be addressed and integrated into the overall system interface requirements. An approach for defining and evaluating the system infrastructures including the testbed currently being used to support the on-going evaluations for the evolutionary Space Station Freedom Data Management System is presented and discussed. Intelligent system technologies discussed include artificial intelligence (real-time replanning and scheduling), high performance computational elements (parallel processors, photonic processors, and neural networks), real-time fault management and control, and system software development tools for rapid prototyping capabilities.

Evaluation of speech recognizers for use in advanced combat helicopter crew station research and development

NASA Technical Reports Server (NTRS)

Simpson, Carol A.

1990-01-01

The U.S. Army Crew Station Research and Development Facility uses vintage 1984 speech recognizers. An evaluation was performed of newer off-the-shelf speech recognition devices to determine whether newer technology performance and capabilities are substantially better than that of the Army's current speech recognizers. The Phonetic Discrimination (PD-100) Test was used to compare recognizer performance in two ambient noise conditions: quiet office and helicopter noise. Test tokens were spoken by males and females and in isolated-word and connected-work mode. Better overall recognition accuracy was obtained from the newer recognizers. Recognizer capabilities needed to support the development of human factors design requirements for speech command systems in advanced combat helicopters are listed.

Acoustic Event Detection and Classification

NASA Astrophysics Data System (ADS)

Temko, Andrey; Nadeu, Climent; Macho, Dušan; Malkin, Robert; Zieger, Christian; Omologo, Maurizio

The human activity that takes place in meeting rooms or classrooms is reflected in a rich variety of acoustic events (AE), produced either by the human body or by objects handled by humans, so the determination of both the identity of sounds and their position in time may help to detect and describe that human activity. Indeed, speech is usually the most informative sound, but other kinds of AEs may also carry useful information, for example, clapping or laughing inside a speech, a strong yawn in the middle of a lecture, a chair moving or a door slam when the meeting has just started. Additionally, detection and classification of sounds other than speech may be useful to enhance the robustness of speech technologies like automatic speech recognition.

CSP: A Multifaceted Hybrid Architecture for Space Computing

NASA Technical Reports Server (NTRS)

Rudolph, Dylan; Wilson, Christopher; Stewart, Jacob; Gauvin, Patrick; George, Alan; Lam, Herman; Crum, Gary Alex; Wirthlin, Mike; Wilson, Alex; Stoddard, Aaron

2014-01-01

Research on the CHREC Space Processor (CSP) takes a multifaceted hybrid approach to embedded space computing. Working closely with the NASA Goddard SpaceCube team, researchers at the National Science Foundation (NSF) Center for High-Performance Reconfigurable Computing (CHREC) at the University of Florida and Brigham Young University are developing hybrid space computers that feature an innovative combination of three technologies: commercial-off-the-shelf (COTS) devices, radiation-hardened (RadHard) devices, and fault-tolerant computing. Modern COTS processors provide the utmost in performance and energy-efficiency but are susceptible to ionizing radiation in space, whereas RadHard processors are virtually immune to this radiation but are more expensive, larger, less energy-efficient, and generations behind in speed and functionality. By featuring COTS devices to perform the critical data processing, supported by simpler RadHard devices that monitor and manage the COTS devices, and augmented with novel uses of fault-tolerant hardware, software, information, and networking within and between COTS devices, the resulting system can maximize performance and reliability while minimizing energy consumption and cost. NASA Goddard has adopted the CSP concept and technology with plans underway to feature flight-ready CSP boards on two upcoming space missions.

A parallel implementation of an off-lattice individual-based model of multicellular populations

NASA Astrophysics Data System (ADS)

Harvey, Daniel G.; Fletcher, Alexander G.; Osborne, James M.; Pitt-Francis, Joe

2015-07-01

As computational models of multicellular populations include ever more detailed descriptions of biophysical and biochemical processes, the computational cost of simulating such models limits their ability to generate novel scientific hypotheses and testable predictions. While developments in microchip technology continue to increase the power of individual processors, parallel computing offers an immediate increase in available processing power. To make full use of parallel computing technology, it is necessary to develop specialised algorithms. To this end, we present a parallel algorithm for a class of off-lattice individual-based models of multicellular populations. The algorithm divides the spatial domain between computing processes and comprises communication routines that ensure the model is correctly simulated on multiple processors. The parallel algorithm is shown to accurately reproduce the results of a deterministic simulation performed using a pre-existing serial implementation. We test the scaling of computation time, memory use and load balancing as more processes are used to simulate a cell population of fixed size. We find approximate linear scaling of both speed-up and memory consumption on up to 32 processor cores. Dynamic load balancing is shown to provide speed-up for non-regular spatial distributions of cells in the case of a growing population.

Scaling and universality in the human voice.

PubMed

Luque, Jordi; Luque, Bartolo; Lacasa, Lucas

2015-04-06

Speech is a distinctive complex feature of human capabilities. In order to understand the physics underlying speech production, in this work, we empirically analyse the statistics of large human speech datasets ranging several languages. We first show that during speech, the energy is unevenly released and power-law distributed, reporting a universal robust Gutenberg-Richter-like law in speech. We further show that such 'earthquakes in speech' show temporal correlations, as the interevent statistics are again power-law distributed. As this feature takes place in the intraphoneme range, we conjecture that the process responsible for this complex phenomenon is not cognitive, but it resides in the physiological (mechanical) mechanisms of speech production. Moreover, we show that these waiting time distributions are scale invariant under a renormalization group transformation, suggesting that the process of speech generation is indeed operating close to a critical point. These results are put in contrast with current paradigms in speech processing, which point towards low dimensional deterministic chaos as the origin of nonlinear traits in speech fluctuations. As these latter fluctuations are indeed the aspects that humanize synthetic speech, these findings may have an impact in future speech synthesis technologies. Results are robust and independent of the communication language or the number of speakers, pointing towards a universal pattern and yet another hint of complexity in human speech. © 2015 The Author(s) Published by the Royal Society. All rights reserved.

Successful development of first-generation laser device; marking China's optoelectronic technology at world class level

NASA Astrophysics Data System (ADS)

1995-04-01

Bell Laboratories has developed the world's first optical information processor. Its core device is a self-excited electrooptical effect apparatus array of symmetric operation. After being developed in the United States, this high-technology device was successfully developed by China's scientists,thus making the fact that China's optoelectronic technology is among the most advanced in the world.

Advanced Physiological Estimation of Cognitive Status. Part 2

DTIC Science & Technology

2011-05-24

Neurofeedback Algorithms and Gaze Controller EEG Sensor System g.USBamp *, ** • internal 24-bit ADC and digital signal processor • 16 channels (expandable...SUBJECT TERMS EEG eye-tracking mental state estimation machine learning Leonard J. Trejo Pacific Development and Technology LLC 999 Commercial St. Palo...fatigue, overload) Technology Transfer Opportunity Technology from PDT – Methods to acquire various physiological signals ( EEG , EOG, EMG, ECG, etc

Digital Collaboration Tools in the Military: Their Historical and Current Status

DTIC Science & Technology

2006-02-16

Writer = online word processor that edits, stores and shares your documents from anywhere. February 16, 2006 31 Recent “ Disruptive ” Technologies Cell...Webcasts Wikis February 16, 2006 32 Now Consider: Disruptive Technologies (1997) becomes Disruptive Innovations in 2003. Military Transformation: Drivers...from http://www.sims.berkeley.edu/how-much-info-2003 Schneiderman, R. (2005). Preparing for the Disruptive Technologies of Tomorrow. http

Speech transport for packet telephony and voice over IP

NASA Astrophysics Data System (ADS)

Baker, Maurice R.

1999-11-01

Recent advances in packet switching, internetworking, and digital signal processing technologies have converged to allow realizable practical implementations of packet telephony systems. This paper provides a tutorial on transmission engineering for packet telephony covering the topics of speech coding/decoding, speech packetization, packet data network transport, and impairments which may negatively impact end-to-end system quality. Particular emphasis is placed upon Voice over Internet Protocol given the current popularity and ubiquity of IP transport.

Sensitivity to pulse phase duration in cochlear implant listeners: Effects of stimulation mode

PubMed Central

Chatterjee, Monita; Kulkarni, Aditya M.

2014-01-01

The objective of this study was to investigate charge-integration at threshold by cochlear implant listeners using pulse train stimuli in different stimulation modes (monopolar, bipolar, tripolar). The results partially confirmed and extended the findings of previous studies conducted in animal models showing that charge-integration depends on the stimulation mode. The primary overall finding was that threshold vs pulse phase duration functions had steeper slopes in monopolar mode and shallower slopes in more spatially restricted modes. While the result was clear-cut in eight users of the Cochlear CorporationTM device, the findings with the six user of the Advanced BionicsTM device who participated were less consistent. It is likely that different stimulation modes excite different neuronal populations and/or sites of excitation on the same neuron (e.g., peripheral process vs central axon). These differences may influence not only charge integration but possibly also temporal dynamics at suprathreshold levels and with more speech-relevant stimuli. Given the present interest in focused stimulation modes, these results have implications for cochlear implant speech processor design and protocols used to map acoustic amplitude to electric stimulation parameters. PMID:25096116

Frequency modulation detection in cochlear implant subjects

NASA Astrophysics Data System (ADS)

Chen, Hongbin; Zeng, Fan-Gang

2004-10-01

Frequency modulation (FM) detection was investigated in acoustic and electric hearing to characterize cochlear-implant subjects' ability to detect dynamic frequency changes and to assess the relative contributions of temporal and spectral cues to frequency processing. Difference limens were measured for frequency upward sweeps, downward sweeps, and sinusoidal FM as a function of standard frequency and modulation rate. In electric hearing, factors including electrode position and stimulation level were also studied. Electric hearing data showed that the difference limen increased monotonically as a function of standard frequency regardless of the modulation type, the modulation rate, the electrode position, and the stimulation level. In contrast, acoustic hearing data showed that the difference limen was nearly a constant as a function of standard frequency. This difference was interpreted to mean that temporal cues are used only at low standard frequencies and at low modulation rates. At higher standard frequencies and modulation rates, the reliance on the place cue is increased, accounting for the better performance in acoustic hearing than for electric hearing with single-electrode stimulation. The present data suggest a speech processing strategy that encodes slow frequency changes using lower stimulation rates than those typically employed by contemporary cochlear-implant speech processors. .

Opening Statements and Speeches. Plenary Session. Papers.

ERIC Educational Resources Information Center

International Federation of Library Associations, The Hague (Netherlands).

Official opening statements, organizational reports, and papers on libraries in a technological world, which were presented at the 1983 International Federation of Library Associations (IFLA) conference include: (1) welcoming addresses by Franz Georg Kaltwasser and Mathilde Berghofer-Weichner; (2) opening speeches by Else Granheim (IFLA president)…

Using Telerehabilitation to Assess Apraxia of Speech in Adults

ERIC Educational Resources Information Center

Hill, Anne Jane; Theodoros, Deborah; Russell, Trevor; Ward, Elizabeth

2009-01-01

Background: Telerehabilitation is the remote delivery of rehabilitation services via information technology and telecommunication systems. There have been a number of studies that have used videoconferencing to assess speech and language skills in people with acquired neurogenic communication disorders. However, few studies have focused on cases…

Bit-parallel arithmetic in a massively-parallel associative processor

NASA Technical Reports Server (NTRS)

Scherson, Isaac D.; Kramer, David A.; Alleyne, Brian D.

1992-01-01

A simple but powerful new architecture based on a classical associative processor model is presented. Algorithms for performing the four basic arithmetic operations both for integer and floating point operands are described. For m-bit operands, the proposed architecture makes it possible to execute complex operations in O(m) cycles as opposed to O(m exp 2) for bit-serial machines. A word-parallel, bit-parallel, massively-parallel computing system can be constructed using this architecture with VLSI technology. The operation of this system is demonstrated for the fast Fourier transform and matrix multiplication.

Using the automata processor for fast pattern recognition in high energy physics experiments. A proof of concept

DOE PAGES

Michael H. L. S. Wang; Cancelo, Gustavo; Green, Christopher; ...

2016-06-25

Here, we explore the Micron Automata Processor (AP) as a suitable commodity technology that can address the growing computational needs of pattern recognition in High Energy Physics (HEP) experiments. A toy detector model is developed for which an electron track confirmation trigger based on the Micron AP serves as a test case. Although primarily meant for high speed text-based searches, we demonstrate a proof of concept for the use of the Micron AP in a HEP trigger application.

Discover Supercomputer 3

NASA Image and Video Library

2017-12-08

The heart of the NASA Center for Climate Simulation (NCCS) is the “Discover” supercomputer. In 2009, NCCS added more than 8,000 computer processors to Discover, for a total of nearly 15,000 processors. Credit: NASA/Pat Izzo To learn more about NCCS go to: www.nasa.gov/topics/earth/features/climate-sim-center.html NASA Goddard Space Flight Center is home to the nation's largest organization of combined scientists, engineers and technologists that build spacecraft, instruments and new technology to study the Earth, the sun, our solar system, and the universe.

Discover Supercomputer 2

NASA Image and Video Library

2017-12-08

The heart of the NASA Center for Climate Simulation (NCCS) is the “Discover” supercomputer. In 2009, NCCS added more than 8,000 computer processors to Discover, for a total of nearly 15,000 processors. Credit: NASA/Pat Izzo To learn more about NCCS go to: www.nasa.gov/topics/earth/features/climate-sim-center.html NASA Goddard Space Flight Center is home to the nation's largest organization of combined scientists, engineers and technologists that build spacecraft, instruments and new technology to study the Earth, the sun, our solar system, and the universe.

Discover Supercomputer 1

NASA Image and Video Library

2017-12-08

The heart of the NASA Center for Climate Simulation (NCCS) is the “Discover” supercomputer. In 2009, NCCS added more than 8,000 computer processors to Discover, for a total of nearly 15,000 processors. Credit: NASA/Pat Izzo To learn more about NCCS go to: www.nasa.gov/topics/earth/features/climate-sim-center.html NASA Goddard Space Flight Center is home to the nation's largest organization of combined scientists, engineers and technologists that build spacecraft, instruments and new technology to study the Earth, the sun, our solar system, and the universe.

Performance and Power Optimization for Cognitive Processor Design Using Deep-Submicron Very Large Scale Integration (VLSI) Technology

DTIC Science & Technology

2010-03-01

DATES COVERED (From - To) October 2008 – October 2009 4 . TITLE AND SUBTITLE PERFORMANCE AND POWER OPTIMIZATION FOR COGNITIVE PROCESSOR DESIGN USING...Computations 2  2.2  Cognitive Models and Algorithms for Intelligent Text Recognition 4   2.2.1 Brain-State-in-a-Box Neural Network Model. 4   2.2.2...The ASIC-style design and synthesis flow for FPU 8  Figure 4 : Screen shots of the final layouts 10  Figure 5: Projected performance and power roadmap

Using the automata processor for fast pattern recognition in high energy physics experiments. A proof of concept

DOE Office of Scientific and Technical Information (OSTI.GOV)

Michael H. L. S. Wang; Cancelo, Gustavo; Green, Christopher

Here, we explore the Micron Automata Processor (AP) as a suitable commodity technology that can address the growing computational needs of pattern recognition in High Energy Physics (HEP) experiments. A toy detector model is developed for which an electron track confirmation trigger based on the Micron AP serves as a test case. Although primarily meant for high speed text-based searches, we demonstrate a proof of concept for the use of the Micron AP in a HEP trigger application.

49 CFR 234.275 - Processor-based systems.

Code of Federal Regulations, 2014 CFR

2014-10-01

... first placed in service after June 6, 2005, which contain new or novel technology, or which provide safety-critical data to a railroad signal or train control system that is governed by part 236, subpart H or I, of this chapter, shall also comply with those requirements. New or novel technology refers to a...

49 CFR 234.275 - Processor-based systems.

Code of Federal Regulations, 2012 CFR

2012-10-01

... first placed in service after June 6, 2005, which contain new or novel technology, or which provide safety-critical data to a railroad signal or train control system that is governed by part 236, subpart H or I, of this chapter, shall also comply with those requirements. New or novel technology refers to a...

49 CFR 234.275 - Processor-based systems.

Code of Federal Regulations, 2013 CFR

2013-10-01

... first placed in service after June 6, 2005, which contain new or novel technology, or which provide safety-critical data to a railroad signal or train control system that is governed by part 236, subpart H or I, of this chapter, shall also comply with those requirements. New or novel technology refers to a...

TheBrain Technologies Corporation: Collapsing the Time to Knowledge.

ERIC Educational Resources Information Center

Misek, Marla

2003-01-01

TheBrain was created to take advantage of the most powerful information processor in existence - the human mind. Explains products of TheBrain Technologies Corporation,, which has developed computer interfaces to help individual users and corporations organize information in ways that make sense to them in the proper context. Describes a…

Compact propane fuel processor for auxiliary power unit application

NASA Astrophysics Data System (ADS)

Dokupil, M.; Spitta, C.; Mathiak, J.; Beckhaus, P.; Heinzel, A.

With focus on mobile applications a fuel cell auxiliary power unit (APU) using liquefied petroleum gas (LPG) is currently being developed at the Centre for Fuel Cell Technology (Zentrum für BrennstoffzellenTechnik, ZBT gGmbH). The system is consisting of an integrated compact and lightweight fuel processor and a low temperature PEM fuel cell for an electric power output of 300 W. This article is presenting the current status of development of the fuel processor which is designed for a nominal hydrogen output of 1 k Wth,H2 within a load range from 50 to 120%. A modular setup was chosen defining a reformer/burner module and a CO-purification module. Based on the performance specifications, thermodynamic simulations, benchmarking and selection of catalysts the modules have been developed and characterised simultaneously and then assembled to the complete fuel processor. Automated operation results in a cold startup time of about 25 min for nominal load and carbon monoxide output concentrations below 50 ppm for steady state and dynamic operation. Also fast transient response of the fuel processor at load changes with low fluctuations of the reformate gas composition have been achieved. Beside the development of the main reactors the transfer of the fuel processor to an autonomous system is of major concern. Hence, concepts for packaging have been developed resulting in a volume of 7 l and a weight of 3 kg. Further a selection of peripheral components has been tested and evaluated regarding to the substitution of the laboratory equipment.

Speech Perception With Combined Electric-Acoustic Stimulation: A Simulation and Model Comparison.

PubMed

Rader, Tobias; Adel, Youssef; Fastl, Hugo; Baumann, Uwe

2015-01-01

The aim of this study is to simulate speech perception with combined electric-acoustic stimulation (EAS), verify the advantage of combined stimulation in normal-hearing (NH) subjects, and then compare it with cochlear implant (CI) and EAS user results from the authors' previous study. Furthermore, an automatic speech recognition (ASR) system was built to examine the impact of low-frequency information and is proposed as an applied model to study different hypotheses of the combined-stimulation advantage. Signal-detection-theory (SDT) models were applied to assess predictions of subject performance without the need to assume any synergistic effects. Speech perception was tested using a closed-set matrix test (Oldenburg sentence test), and its speech material was processed to simulate CI and EAS hearing. A total of 43 NH subjects and a customized ASR system were tested. CI hearing was simulated by an aurally adequate signal spectrum analysis and representation, the part-tone-time-pattern, which was vocoded at 12 center frequencies according to the MED-EL DUET speech processor. Residual acoustic hearing was simulated by low-pass (LP)-filtered speech with cutoff frequencies 200 and 500 Hz for NH subjects and in the range from 100 to 500 Hz for the ASR system. Speech reception thresholds were determined in amplitude-modulated noise and in pseudocontinuous noise. Previously proposed SDT models were lastly applied to predict NH subject performance with EAS simulations. NH subjects tested with EAS simulations demonstrated the combined-stimulation advantage. Increasing the LP cutoff frequency from 200 to 500 Hz significantly improved speech reception thresholds in both noise conditions. In continuous noise, CI and EAS users showed generally better performance than NH subjects tested with simulations. In modulated noise, performance was comparable except for the EAS at cutoff frequency 500 Hz where NH subject performance was superior. The ASR system showed similar behavior to NH subjects despite a positive signal-to-noise ratio shift for both noise conditions, while demonstrating the synergistic effect for cutoff frequencies ≥300 Hz. One SDT model largely predicted the combined-stimulation results in continuous noise, while falling short of predicting performance observed in modulated noise. The presented simulation was able to demonstrate the combined-stimulation advantage for NH subjects as observed in EAS users. Only NH subjects tested with EAS simulations were able to take advantage of the gap listening effect, while CI and EAS user performance was consistently degraded in modulated noise compared with performance in continuous noise. The application of ASR systems seems feasible to assess the impact of different signal processing strategies on speech perception with CI and EAS simulations. In continuous noise, SDT models were largely able to predict the performance gain without assuming any synergistic effects, but model amendments are required to explain the gap listening effect in modulated noise.

Recognition and localization of speech by adult cochlear implant recipients wearing a digital hearing aid in the nonimplanted ear (bimodal hearing).

PubMed

Potts, Lisa G; Skinner, Margaret W; Litovsky, Ruth A; Strube, Michael J; Kuk, Francis

2009-06-01

The use of bilateral amplification is now common clinical practice for hearing aid users but not for cochlear implant recipients. In the past, most cochlear implant recipients were implanted in one ear and wore only a monaural cochlear implant processor. There has been recent interest in benefits arising from bilateral stimulation that may be present for cochlear implant recipients. One option for bilateral stimulation is the use of a cochlear implant in one ear and a hearing aid in the opposite nonimplanted ear (bimodal hearing). This study evaluated the effect of wearing a cochlear implant in one ear and a digital hearing aid in the opposite ear on speech recognition and localization. A repeated-measures correlational study was completed. Nineteen adult Cochlear Nucleus 24 implant recipients participated in the study. The participants were fit with a Widex Senso Vita 38 hearing aid to achieve maximum audibility and comfort within their dynamic range. Soundfield thresholds, loudness growth, speech recognition, localization, and subjective questionnaires were obtained six-eight weeks after the hearing aid fitting. Testing was completed in three conditions: hearing aid only, cochlear implant only, and cochlear implant and hearing aid (bimodal). All tests were repeated four weeks after the first test session. Repeated-measures analysis of variance was used to analyze the data. Significant effects were further examined using pairwise comparison of means or in the case of continuous moderators, regression analyses. The speech-recognition and localization tasks were unique, in that a speech stimulus presented from a variety of roaming azimuths (140 degree loudspeaker array) was used. Performance in the bimodal condition was significantly better for speech recognition and localization compared to the cochlear implant-only and hearing aid-only conditions. Performance was also different between these conditions when the location (i.e., side of the loudspeaker array that presented the word) was analyzed. In the bimodal condition, the speech-recognition and localization tasks were equal regardless of which side of the loudspeaker array presented the word, while performance was significantly poorer for the monaural conditions (hearing aid only and cochlear implant only) when the words were presented on the side with no stimulation. Binaural loudness summation of 1-3 dB was seen in soundfield thresholds and loudness growth in the bimodal condition. Measures of the audibility of sound with the hearing aid, including unaided thresholds, soundfield thresholds, and the Speech Intelligibility Index, were significant moderators of speech recognition and localization. Based on the questionnaire responses, participants showed a strong preference for bimodal stimulation. These findings suggest that a well-fit digital hearing aid worn in conjunction with a cochlear implant is beneficial to speech recognition and localization. The dynamic test procedures used in this study illustrate the importance of bilateral hearing for locating, identifying, and switching attention between multiple speakers. It is recommended that unilateral cochlear implant recipients, with measurable unaided hearing thresholds, be fit with a hearing aid.

Using Speech Recognition to Enhance the Tongue Drive System Functionality in Computer Access

PubMed Central

Huo, Xueliang; Ghovanloo, Maysam

2013-01-01

Tongue Drive System (TDS) is a wireless tongue operated assistive technology (AT), which can enable people with severe physical disabilities to access computers and drive powered wheelchairs using their volitional tongue movements. TDS offers six discrete commands, simultaneously available to the users, for pointing and typing as a substitute for mouse and keyboard in computer access, respectively. To enhance the TDS performance in typing, we have added a microphone, an audio codec, and a wireless audio link to its readily available 3-axial magnetic sensor array, and combined it with a commercially available speech recognition software, the Dragon Naturally Speaking, which is regarded as one of the most efficient ways for text entry. Our preliminary evaluations indicate that the combined TDS and speech recognition technologies can provide end users with significantly higher performance than using each technology alone, particularly in completing tasks that require both pointing and text entry, such as web surfing. PMID:22255801

Automatic Speech Recognition from Neural Signals: A Focused Review.

PubMed

Herff, Christian; Schultz, Tanja

2016-01-01

Speech interfaces have become widely accepted and are nowadays integrated in various real-life applications and devices. They have become a part of our daily life. However, speech interfaces presume the ability to produce intelligible speech, which might be impossible due to either loud environments, bothering bystanders or incapabilities to produce speech (i.e., patients suffering from locked-in syndrome). For these reasons it would be highly desirable to not speak but to simply envision oneself to say words or sentences. Interfaces based on imagined speech would enable fast and natural communication without the need for audible speech and would give a voice to otherwise mute people. This focused review analyzes the potential of different brain imaging techniques to recognize speech from neural signals by applying Automatic Speech Recognition technology. We argue that modalities based on metabolic processes, such as functional Near Infrared Spectroscopy and functional Magnetic Resonance Imaging, are less suited for Automatic Speech Recognition from neural signals due to low temporal resolution but are very useful for the investigation of the underlying neural mechanisms involved in speech processes. In contrast, electrophysiologic activity is fast enough to capture speech processes and is therefor better suited for ASR. Our experimental results indicate the potential of these signals for speech recognition from neural data with a focus on invasively measured brain activity (electrocorticography). As a first example of Automatic Speech Recognition techniques used from neural signals, we discuss the Brain-to-text system.

Satellite on-board real-time SAR processor prototype

NASA Astrophysics Data System (ADS)

Bergeron, Alain; Doucet, Michel; Harnisch, Bernd; Suess, Martin; Marchese, Linda; Bourqui, Pascal; Desnoyers, Nicholas; Legros, Mathieu; Guillot, Ludovic; Mercier, Luc; Châteauneuf, François

2017-11-01

A Compact Real-Time Optronic SAR Processor has been successfully developed and tested up to a Technology Readiness Level of 4 (TRL4), the breadboard validation in a laboratory environment. SAR, or Synthetic Aperture Radar, is an active system allowing day and night imaging independent of the cloud coverage of the planet. The SAR raw data is a set of complex data for range and azimuth, which cannot be compressed. Specifically, for planetary missions and unmanned aerial vehicle (UAV) systems with limited communication data rates this is a clear disadvantage. SAR images are typically processed electronically applying dedicated Fourier transformations. This, however, can also be performed optically in real-time. Originally the first SAR images were optically processed. The optical Fourier processor architecture provides inherent parallel computing capabilities allowing real-time SAR data processing and thus the ability for compression and strongly reduced communication bandwidth requirements for the satellite. SAR signal return data are in general complex data. Both amplitude and phase must be combined optically in the SAR processor for each range and azimuth pixel. Amplitude and phase are generated by dedicated spatial light modulators and superimposed by an optical relay set-up. The spatial light modulators display the full complex raw data information over a two-dimensional format, one for the azimuth and one for the range. Since the entire signal history is displayed at once, the processor operates in parallel yielding real-time performances, i.e. without resulting bottleneck. Processing of both azimuth and range information is performed in a single pass. This paper focuses on the onboard capabilities of the compact optical SAR processor prototype that allows in-orbit processing of SAR images. Examples of processed ENVISAT ASAR images are presented. Various SAR processor parameters such as processing capabilities, image quality (point target analysis), weight and size are reviewed.

Smart Sensors: Why and when the origin was and why and where the future will be

NASA Astrophysics Data System (ADS)

Corsi, C.

2013-12-01

Smart Sensors is a technique developed in the 70's when the processing capabilities, based on readout integrated with signal processing, was still far from the complexity needed in advanced IR surveillance and warning systems, because of the enormous amount of noise/unwanted signals emitted by operating scenario especially in military applications. The Smart Sensors technology was kept restricted within a close military environment exploding in applications and performances in the 90's years thanks to the impressive improvements in the integrated signal read-out and processing achieved by CCD-CMOS technologies in FPA. In fact the rapid advances of "very large scale integration" (VLSI) processor technology and mosaic EO detector array technology allowed to develop new generations of Smart Sensors with much improved signal processing by integrating microcomputers and other VLSI signal processors. inside the sensor structure achieving some basic functions of living eyes (dynamic stare, non-uniformity compensation, spatial and temporal filtering). New and future technologies (Nanotechnology, Bio-Organic Electronics, Bio-Computing) are lightning a new generation of Smart Sensors extending the Smartness from the Space-Time Domain to Spectroscopic Functional Multi-Domain Signal Processing. History and future forecasting of Smart Sensors will be reported.

An ultra-compact processor module based on the R3000

NASA Astrophysics Data System (ADS)

Mullenhoff, D. J.; Kaschmitter, J. L.; Lyke, J. C.; Forman, G. A.

1992-08-01

Viable high density packaging is of critical importance for future military systems, particularly space borne systems which require minimum weight and size and high mechanical integrity. A leading, emerging technology for high density packaging is multi-chip modules (MCM). During the 1980's, a number of different MCM technologies have emerged. In support of Strategic Defense Initiative Organization (SDIO) programs, Lawrence Livermore National Laboratory (LLNL) has developed, utilized, and evaluated several different MCM technologies. Prior LLNL efforts include modules developed in 1986, using hybrid wafer scale packaging, which are still operational in an Air Force satellite mission. More recent efforts have included very high density cache memory modules, developed using laser pantography. As part of the demonstration effort, LLNL and Phillips Laboratory began collaborating in 1990 in the Phase 3 Multi-Chip Module (MCM) technology demonstration project. The goal of this program was to demonstrate the feasibility of General Electric's (GE) High Density Interconnect (HDI) MCM technology. The design chosen for this demonstration was the processor core for a MIPS R3000 based reduced instruction set computer (RISC), which has been described previously. It consists of the R3000 microprocessor, R3010 floating point coprocessor and 128 Kbytes of cache memory.

Evaluation of Brine Processing Technologies for Spacecraft Wastewater

NASA Technical Reports Server (NTRS)

Shaw, Hali L.; Flynn, Michael; Wisniewski, Richard; Lee, Jeffery; Jones, Harry; Delzeit, Lance; Shull, Sarah; Sargusingh, Miriam; Beeler, David; Howard, Jeanie;

Clinicians' perspectives of therapeutic alliance in face-to-face and telepractice speech-language pathology sessions.

PubMed

Freckmann, Anneka; Hines, Monique; Lincoln, Michelle

2017-06-01

To investigate the face validity of a measure of therapeutic alliance for paediatric speech-language pathology and to determine whether a difference exists in therapeutic alliance reported by speech-language pathologists (SLPs) conducting face-to-face sessions, compared with telepractice SLPs or in their ratings of confidence with technology. SLPs conducting telepractice (n = 14) or face-to-face therapy (n = 18) completed an online survey which included the Therapeutic Alliance Scales for Children - Revised (TASC-r) (Therapist Form) to rate clinicians' perceptions of rapport with up to three clients. Participants also reported their overall perception of rapport with each client and their comfort with technology. There was a strong correlation between TASC-r total scores and overall ratings of rapport, providing preliminary evidence of TASC-r face validity. There was no significant difference between TASC-r scores for telepractice and face-to-face therapy (p = 0.961), nor face-to-face and telepractice SLPs' confidence with familiar (p = 0.414) or unfamiliar technology (p = 0.780). The TASC-r may be a promising tool for measuring therapeutic alliance in speech-language pathology. Telepractice does not appear to have a negative effect on rapport between SLPs and paediatric clients. Future research is required to identify how SLPs develop rapport in telepractice.

A low power biomedical signal processor ASIC based on hardware software codesign.

PubMed

Nie, Z D; Wang, L; Chen, W G; Zhang, T; Zhang, Y T

2009-01-01

A low power biomedical digital signal processor ASIC based on hardware and software codesign methodology was presented in this paper. The codesign methodology was used to achieve higher system performance and design flexibility. The hardware implementation included a low power 32bit RISC CPU ARM7TDMI, a low power AHB-compatible bus, and a scalable digital co-processor that was optimized for low power Fast Fourier Transform (FFT) calculations. The co-processor could be scaled for 8-point, 16-point and 32-point FFTs, taking approximate 50, 100 and 150 clock circles, respectively. The complete design was intensively simulated using ARM DSM model and was emulated by ARM Versatile platform, before conducted to silicon. The multi-million-gate ASIC was fabricated using SMIC 0.18 microm mixed-signal CMOS 1P6M technology. The die area measures 5,000 microm x 2,350 microm. The power consumption was approximately 3.6 mW at 1.8 V power supply and 1 MHz clock rate. The power consumption for FFT calculations was less than 1.5 % comparing with the conventional embedded software-based solution.

Assessing the Progress of Trapped-Ion Processors Towards Fault-Tolerant Quantum Computation

NASA Astrophysics Data System (ADS)

Bermudez, A.; Xu, X.; Nigmatullin, R.; O'Gorman, J.; Negnevitsky, V.; Schindler, P.; Monz, T.; Poschinger, U. G.; Hempel, C.; Home, J.; Schmidt-Kaler, F.; Biercuk, M.; Blatt, R.; Benjamin, S.; Müller, M.

2017-10-01

A quantitative assessment of the progress of small prototype quantum processors towards fault-tolerant quantum computation is a problem of current interest in experimental and theoretical quantum information science. We introduce a necessary and fair criterion for quantum error correction (QEC), which must be achieved in the development of these quantum processors before their sizes are sufficiently big to consider the well-known QEC threshold. We apply this criterion to benchmark the ongoing effort in implementing QEC with topological color codes using trapped-ion quantum processors and, more importantly, to guide the future hardware developments that will be required in order to demonstrate beneficial QEC with small topological quantum codes. In doing so, we present a thorough description of a realistic trapped-ion toolbox for QEC and a physically motivated error model that goes beyond standard simplifications in the QEC literature. We focus on laser-based quantum gates realized in two-species trapped-ion crystals in high-optical aperture segmented traps. Our large-scale numerical analysis shows that, with the foreseen technological improvements described here, this platform is a very promising candidate for fault-tolerant quantum computation.

Difficulty understanding speech in noise by the hearing impaired: underlying causes and technological solutions.

PubMed

Healy, Eric W; Yoho, Sarah E

2016-08-01

A primary complaint of hearing-impaired individuals involves poor speech understanding when background noise is present. Hearing aids and cochlear implants often allow good speech understanding in quiet backgrounds. But hearing-impaired individuals are highly noise intolerant, and existing devices are not very effective at combating background noise. As a result, speech understanding in noise is often quite poor. In accord with the significance of the problem, considerable effort has been expended toward understanding and remedying this issue. Fortunately, our understanding of the underlying issues is reasonably good. In sharp contrast, effective solutions have remained elusive. One solution that seems promising involves a single-microphone machine-learning algorithm to extract speech from background noise. Data from our group indicate that the algorithm is capable of producing vast increases in speech understanding by hearing-impaired individuals. This paper will first provide an overview of the speech-in-noise problem and outline why hearing-impaired individuals are so noise intolerant. An overview of our approach to solving this problem will follow.

A dissociation of objective and subjective workload measures in assessing the impact of speech controls in advanced helicopters

NASA Technical Reports Server (NTRS)

Vidulich, Michael A.; Bortolussi, Michael R.

1988-01-01

Among the new technologies that are expected to aid helicopter designers are speech controls. Proponents suggest that speech controls could reduce the potential for manual control overloads and improve time-sharing performance in environments that have heavy demands for manual control. This was tested in a simulation of an advanced single-pilot, scout/attack helicopter. Objective performance indicated that the speech controls were effective in decreasing the interference of discrete responses during moments of heavy flight control activity. However, subjective ratings indicated that the use of speech controls required extra effort to speak precisely and to attend to feedback. Although the operational reliability of speech controls must be improved, the present results indicate that reliable speech controls could enhance the time-sharing efficiency of helicopter pilots. Furthermore, the results demonstrated the importance of using multiple assessment techniques to completely assess a task. Neither the objective nor the subjective measures alone provided complete information. It was the contrast between the measures that was most informative.

[Creating language model of the forensic medicine domain for developing a autopsy recording system by automatic speech recognition].

PubMed

Niijima, H; Ito, N; Ogino, S; Takatori, T; Iwase, H; Kobayashi, M

2000-11-01

For the purpose of practical use of speech recognition technology for recording of forensic autopsy, a language model of the speech recording system, specialized for the forensic autopsy, was developed. The language model for the forensic autopsy by applying 3-gram model was created, and an acoustic model for Japanese speech recognition by Hidden Markov Model in addition to the above were utilized to customize the speech recognition engine for forensic autopsy. A forensic vocabulary set of over 10,000 words was compiled and some 300,000 sentence patterns were made to create the forensic language model, then properly mixing with a general language model to attain high exactitude. When tried by dictating autopsy findings, this speech recognition system was proved to be about 95% of recognition rate that seems to have reached to the practical usability in view of speech recognition software, though there remains rooms for improving its hardware and application-layer software.

Advances in natural language processing.

PubMed

Hirschberg, Julia; Manning, Christopher D

2015-07-17

Natural language processing employs computational techniques for the purpose of learning, understanding, and producing human language content. Early computational approaches to language research focused on automating the analysis of the linguistic structure of language and developing basic technologies such as machine translation, speech recognition, and speech synthesis. Today's researchers refine and make use of such tools in real-world applications, creating spoken dialogue systems and speech-to-speech translation engines, mining social media for information about health or finance, and identifying sentiment and emotion toward products and services. We describe successes and challenges in this rapidly advancing area. Copyright © 2015, American Association for the Advancement of Science.

Syntactic error modeling and scoring normalization in speech recognition

NASA Technical Reports Server (NTRS)

Olorenshaw, Lex

1991-01-01

The objective was to develop the speech recognition system to be able to detect speech which is pronounced incorrectly, given that the text of the spoken speech is known to the recognizer. Research was performed in the following areas: (1) syntactic error modeling; (2) score normalization; and (3) phoneme error modeling. The study into the types of errors that a reader makes will provide the basis for creating tests which will approximate the use of the system in the real world. NASA-Johnson will develop this technology into a 'Literacy Tutor' in order to bring innovative concepts to the task of teaching adults to read.

Hybrid fuel cell/diesel generation total energy system, part 2

NASA Astrophysics Data System (ADS)

Blazek, C. F.

1982-11-01

Meeting the Goldstone Deep Space Communications Complex (DGSCC) electrical and thermal requirements with the existing system was compared with using fuel cells. Fuel cell technology selection was based on a 1985 time frame for installation. The most cost-effective fuel feedstock for fuel cell application was identified. Fuels considered included diesel oil, natural gas, methanol and coal. These fuel feedstocks were considered not only on the cost and efficiency of the fuel conversion process, but also on complexity and integration of the fuel processor on system operation and thermal energy availability. After a review of fuel processor technology, catalytic steam reformer technology was selected based on the ease of integration and the economics of hydrogen production. The phosphoric acid fuel cell was selected for application at the GDSCC due to its commercial readiness for near term application. Fuel cell systems were analyzed for both natural gas and methanol feedstock. The subsequent economic analysis indicated that a natural gas fueled system was the most cost effective of the cases analyzed.

Hybrid fuel cell/diesel generation total energy system, part 2

NASA Technical Reports Server (NTRS)

Blazek, C. F.

1982-01-01

Meeting the Goldstone Deep Space Communications Complex (DGSCC) electrical and thermal requirements with the existing system was compared with using fuel cells. Fuel cell technology selection was based on a 1985 time frame for installation. The most cost-effective fuel feedstock for fuel cell application was identified. Fuels considered included diesel oil, natural gas, methanol and coal. These fuel feedstocks were considered not only on the cost and efficiency of the fuel conversion process, but also on complexity and integration of the fuel processor on system operation and thermal energy availability. After a review of fuel processor technology, catalytic steam reformer technology was selected based on the ease of integration and the economics of hydrogen production. The phosphoric acid fuel cell was selected for application at the GDSCC due to its commercial readiness for near term application. Fuel cell systems were analyzed for both natural gas and methanol feedstock. The subsequent economic analysis indicated that a natural gas fueled system was the most cost effective of the cases analyzed.

Legal Issues and Computer Use by School-Based Audiologists and Speech-Language Pathologists.

ERIC Educational Resources Information Center

Wynne, Michael K.; Hurst, David S.

1995-01-01

This article reviews ethical and legal issues regarding school-based integration and application of technologies, particularly when used by speech-language pathologists and audiologists. Four issues are addressed: (1) software copyright and licensed use; (2) information access and the right to privacy; (3) computer-assisted or…

Expanding Use of Telepractice in Speech-Language Pathology and Audiology

ERIC Educational Resources Information Center

Edwards, Marge; Stredler-Brown, Arlene; Houston, K. Todd

2012-01-01

Recent advances in videoconferencing technology have resulted in a substantial increase in the use of live videoconferencing--referred to here as telepractice--to diagnose and treat speech, language, and hearing disorders. There is growing support from professional organizations for use of this service delivery model, as videoconferencing…

The Forces Restructuring Our Future and Outdoor Recreation: Transcription of Keynote Speech.

ERIC Educational Resources Information Center

Feather, Frank

This futurist keynote speech of the National Conference for Outdoor Leaders addresses the social, technological, economic, and political forces that are restructuring the world. The concept of geostrategic thinking has the components of global thinking, futuristic thinking, and seeking opportunities. Important developments include: (1) wealth will…

Assessing Disordered Speech and Voice in Parkinson's Disease: A Telerehabilitation Application

ERIC Educational Resources Information Center

Constantinescu, Gabriella; Theodoros, Deborah; Russell, Trevor; Ward, Elizabeth; Wilson, Stephen; Wootton, Richard

2010-01-01

Background: Patients with Parkinson's disease face numerous access barriers to speech pathology services for appropriate assessment and treatment. Telerehabilitation is a possible solution to this problem, whereby rehabilitation services may be delivered to the patient at a distance, via telecommunication and information technologies. A number of…

Connecting Intonation Labels to Mathematical Descriptions of Fundamental Frequency

ERIC Educational Resources Information Center

Grabe, Esther; Kochanski, Greg; Coleman, John

2007-01-01

The mathematical models of intonation used in speech technology are often inaccessible to linguists. By the same token, phonological descriptions of intonation are rarely used by speech technologists, as they cannot be implemented directly in applications. Consequently, these research communities do not benefit much from each other's insights. In…