Tesis sobre el tema "Speech coding"
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Abboud, Karim. "Wideband CELP speech coding". Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56805.
Texto completothe first approach considers the quantization of Liner Predictive Coding (LPC) parameters and uses a three way split vector quantization. Both scalar and vector quantization are initially studied; results show that, with adequate codebook training, the second method generates better results while using a fewer number of bits. Nevertheless, the use of vector quantizers remain highly complex in terms of memory and number of computations. A new quantization scheme, split vector quantization (split VQ), is investigated to overcome this complexity problem. Using a new weighted distance measure as a selection criterion for split VQ, the average spectral distortion is significantly reduced to match the results obtained with scalar quantizers.
The second approach introduces a new pitch predictor with an increased temporal resolution for periodicity. This new technique has the advantage of maintaining the same quality obtained with conventional multiple coefficient predictors at a reduced bit rate. Furthermore, the conventional CELP noise weighting filter is modified to allow more freedom and better accuracy in the modeling of both tilt and formant structures. Throughout this process, different noise weighting schemes are evaluated and the results show that the new filter greatly contributes in solving the problem of high frequency distortion.
The final wideband CELP coder is operational at 11.7 kbits/s and generates a high perceptual quality of the reconstructed speech using the fractional pitch predictor and the new perceptual noise weighting filter.
Sturt, Christian. "Pitch synchronous speech coding techniques". Thesis, University of Surrey, 2003. http://epubs.surrey.ac.uk/843327/.
Texto completoKaouri, Hussein Ali. "Speech coding using vector quantisation". Thesis, Queen's University Belfast, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.356934.
Texto completoKritzinger, Carl. "Low bit rate speech coding". Thesis, Stellenbosch : University of Stellenbosch, 2006. http://hdl.handle.net/10019.1/2078.
Texto completoDespite enormous advances in digital communication, the voice is still the primary tool with which people exchange ideas. However, uncompressed digital speech tends to require prohibitively high data rates (upward of 64kbps), making it impractical for many applications. Speech coding is the process of reducing the data rate of digital voice to manageable levels. Parametric speech coders or vocoders utilise a-priori information about the mechanism by which speech is produced in order to achieve extremely efficient compression of speech signals (as low as 1 kbps). The greater part of this thesis comprises an investigation into parametric speech coding. This consisted of a review of the mathematical and heuristic tools used in parametric speech coding, as well as the implementation of an accepted standard algorithm for parametric voice coding. In order to examine avenues of improvement for the existing vocoders, we examined some of the mathematical structure underlying parametric speech coding. Following on from this, we developed a novel approach to parametric speech coding which obtained promising results under both objective and subjective evaluation. An additional contribution by this thesis was the comparative subjective evaluation of the effect of parametric speech coding on English and Xhosa speech. We investigated the performance of two different encoding algorithms on the two languages.
Burnett, I. S. "Hybrid techniques for speech coding". Thesis, University of Bath, 1992. https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.317353.
Texto completoAl-Naimi, Khaldoon Taha. "Advanced speech processing and coding techniques". Thesis, University of Surrey, 2002. http://epubs.surrey.ac.uk/843488/.
Texto completoZhao, David Yuheng. "Model Based Speech Enhancement and Coding". Doctoral thesis, Stockholm : Kungliga Tekniska högskolan, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-4412.
Texto completoKatugampala, Nilantha N. "Multimode speech coding below 6 kbps". Thesis, University of Surrey, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.365141.
Texto completoGreen, Richard C. "Walsh based cepstra for speech coding". Thesis, King's College London (University of London), 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392848.
Texto completoOoi, James M. 1970. "Application of wavelets to speech coding". Thesis, Massachusetts Institute of Technology, 1993. http://hdl.handle.net/1721.1/12340.
Texto completoZolfaghari, Parham Seyed. "Sinusoidal model based segmental speech coding". Thesis, University of Cambridge, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.621177.
Texto completoMason, Michael. "Hybrid coding of speech and audio signals". Thesis, Queensland University of Technology, 2001.
Buscar texto completoBatri, Nadim. "Robust spectral parameter coding in speech processing". Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0005/MQ43996.pdf.
Texto completoAsenstorfer, John A. "Source-channel coding for CELP speech coders /". Title page, contents and abstract only, 1994. http://web4.library.adelaide.edu.au/theses/09PH/09pha816.pdf.
Texto completoSoong, Michael. "Predictive split vector quantization for speech coding". Thesis, McGill University, 1994. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=68054.
Texto completoSummation Product Codes (SPCs) are a family of structured vector quantizers that circumvent the complexity obstacle. The performance of SPC vector quantizers can be traded off against their storage and encoding complexity. Besides the complexity factors, the design algorithm can also affect the performance of the quantizer. The conventional generalized Lloyd's algorithm (GLA) generates sub-optimal codebooks. For particular SPC such as multistage VQ, the GLA is applied to design the stage codebooks stage-by-stage. Joint design algorithms on the other hand update all the stage codebooks simultaneously.
In this thesis, a general formulation and an algorithm solution to the joint codebook design problem is provided for the SPCs. The key to this algorithm is that every PC has a reference product codebook which minimizes the overall distortion. This joint design algorithm is tested with a novel SPC, namely "Predictive Split VQ (PSVQ)".
VQ of speech Line Spectral Frequencies (LSF's) using PSVQ is also presented. A result in this work is that PSVQ, designed using the joint codebook design algorithm requires only 20 bits/frame(20 ms) for transparent coding of a 10$ sp{ rm th}$ order LSF's parameters.
Grass, John. "Quantization of predictor coefficients in speech coding". Thesis, McGill University, 1990. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=60067.
Texto completoScalar quantization is the first approach evaluated. Results show that Line Spectral Frequencies require significantly fewer bits than reflection coefficients for comparable performance. The second approach investigated is the use of vector-scalar quantization. In the first stage, vector quantization is performed. The second stage consists of a bank of scalar quantizers which code the vector errors between the original LPC coefficients and the components of the vector of the quantized coefficients.
The approach is to couple the vector and scalar quantization stages. Every codebook vector is compared to the original LPC coefficient vector to produce error vectors. The second innovation into vector-scalar quantization is the incorporation of a small adaptive codebook to the large fixed codebook. Frame-to-frame correlation of the LPC coefficients is exploited at no extra cost in bits.
The performance of the vector-scalar quantization using the two new techniques is better than that of the scalar coding techniques currently used in conventional LPC coders.
Maroun, Nabih. "Toll-quality speech coding at 8 kbs". Thesis, McGill University, 1993. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56802.
Texto completoSuddle, Muhammad Riaz. "Speech coding in private and broadcast networks". Thesis, University of Surrey, 1996. http://epubs.surrey.ac.uk/1019/.
Texto completoOberhofer, Robert. "Pitch adaptive variable bitrate CELP speech coding". Thesis, University of Ulster, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.264811.
Texto completoThorpe, T. F. "Performance bounds for digital coding of speech". Thesis, University of Cambridge, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.234070.
Texto completoGant, Nicolas Roland Noel. "The linear predictive coding of mask speech". Thesis, University of Southampton, 1986. https://eprints.soton.ac.uk/52261/.
Texto completoDeloche, François. "Short time-scale efficient coding of speech". Thesis, Paris, EHESS, 2019. http://www.theses.fr/2019EHES0142.
Texto completoCochlear frequency selectivity is known to reflect the overall statistical structure of speech, in line with the hypothesis that low-level sensory processing provides efficient codes for information contained in natural stimuli. Speech signals, however, possess a complex structure, even on short-time scales, as a result of the diversity of acoustic factors involved in the generation of speech. This rich structure means that advanced coding schemes based on a nonlinear representation of speech sounds could provide more efficient codes. The first step in finding efficient strategies is to describe the statistical structure of speech at a fine level — at the level of phonemes or even finer at the level of acoustic events. In this thesis, I use a parametric approach to explore the fine-grained statistical structure of speech. The goal of this method is to find the sparsest representation of speech sounds among a family of dictionaries of Gabor filters whose frequency selectivity follows different power laws in the high frequency range 1-8kHz. I motivate the use of Gabor filters for the search of sparse time-frequency representations of speech signals, and I show that the dictionary method has a formal link with previous work based on Independent Component Analysis (ICA). The acoustic factors that affect the power law associated with the sparsest decomposition can be inferred from the analyses of synthetic and real data. The results suggest that an efficient speech coding strategy is to reduce frequency selectivity with sound intensity level, reflecting the nonlinear behavior of the cochlea
Hoyle, Robert D. (Robert Douglas) Carleton University Dissertation Engineering Electrical. "Digital speech coding for land mobile radio". Ottawa, 1986.
Buscar texto completoGreenwood, Andrew Richard. "Articulatory speech synthesis". Thesis, University of Liverpool, 1993. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.386773.
Texto completoVarga, A. P. "Multipulse excited linear predictive analysis in speech coding and constructive speech synthesis". Thesis, University of Cambridge, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.372909.
Texto completoLeong, Michael. "Representing voiced speech using prototype waveform interpolation for low-rate speech coding". Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56796.
Texto completoIn examining the PWI method, it was found that although the method generally works very well there are occasional sections of the reconstructed voiced speech where audible distortion can be heard, even when the prototypes are not quantized. The research undertaken in this thesis focuses on the fundamental principles behind modelling voiced speech using PWI instead of focusing on bit allocation for encoding the prototypes. Problems in the PWI method are found that may be have been overlooked as encoding error if full encoding were implemented.
Kleijn uses PWI to represent voiced sections of the excitation signal which is the residual obtained after the removal of short-term redundancies by a linear predictive filter. The problem with this method is that when the PWI reconstructed excitation is passed through the inverse filter to synthesize the speech undesired effects occur due to the time-varying nature of the filter. The reconstructed speech may have undesired envelope variations which result in audible warble.
This thesis proposes an energy fixup to smoothen the synthesized speech envelope when the interpolation procedure fails to provide the smooth linear result that is desired. Further investigation, however, leads to the final proposal in this thesis that PWI should he performed on the clean speech signal instead of the excitation to achieve consistently reliable results for all voiced frames.
Accardi, Anthony J. (Anthony Joseph) 1976. "A modular approach to speech enhancement with an application to speech coding". Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/9976.
Texto completoIncludes bibliographical references (p. 98-101).
by Anthony J. Accardi.
B.S.
M.Eng.
Islam, Tamanna. "Interpolation of linear prediction coefficients for speech coding". Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0034/MQ64229.pdf.
Texto completoTrinkaus, Trevor R. "Perceptual coding of audio and diverse speech signals". Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/13883.
Texto completoLoo, James H. Y. (James Hung Yan). "Intraframe and interframe coding of speech spectral parameters". Thesis, McGill University, 1996. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=24065.
Texto completoBecause speech is quasi-stationary, interframe coding methods such as predictive SVQ (PSVQ) can exploit the correlation between adjacent LSF vectors. Nonlinear PSVQ (NPSVQ) is introduced in which a nonparametric and nonlinear predictor replaces the linear predictor used in PSVQ. Regardless of predictor type, PSVQ garners a performance gain of 5-7 bits/frame over SVQ. By interleaving intraframe SVQ with PSVQ, error propagation is limited to at most one adjacent frame. At an overall bit rate of about 21 bits/frame, NPSVQ can provide similar coding quality as intraframe SVQ at 24 bits/frame (an average gain of 3 bits/frame). The particular form of nonlinear prediction we use incurs virtually no additional encoding computational complexity. Voicing classification is used in classified NPSVQ (CNPSVQ) to obtain an additional average gain of 1 bit/frame for unvoiced frames. Furthermore, switched-adaptive predictive SVQ (SA-PSVQ) provides an improvement of 1 bit/frame over PSVQ, or 6-8 bits/frame over SVQ, but error propagation increases to 3-7 frames. We have verified our comparative performance results using subjective listening tests.
Ramachandran, Ravi P. "Pitch filtering in adaptive predictive coding of speech". Thesis, McGill University, 1986. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=65345.
Texto completoRoy, Guylain. "Low-rate analysis-by-synthesis wideband speech coding". Thesis, McGill University, 1990. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=59643.
Texto completoThe study consists of three stages. First, aspects of wideband spectral envelope modeling using Line Spectral Frequencies (LSF's) are studied. Then, the underlying coder structure is derived from a basic Residual Excited Linear Predictive coder (RELP). This structure is enhanced by the addition of a pitch prediction stage, and by the development of full-band and split-band pitch parameter optimization procedures. These procedures are then applied to an Code Excited Linear Prediction (CELP) model. Finally, the performance of full-band and split-band CELP structures are compared.
Chahine, Gebrael. "Pitch modelling for speech coding at 4.8 kbitss". Thesis, McGill University, 1993. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=69724.
Texto completoA multi-tap LTP outperforms a single-tap LTP, but at the expense of a greater number of bits. A single-tap LTP can be improved by increasing the time resolution of the LTP. This results in a fractional delay LTP, which produces a significant increase in prediction gain and perceived periodicity at the cost of more bits, but less than for the multi-tap case.
The first new approach in this work is to use a pseudo-three-tap pitch filter with one or two degrees of freedom of the predictor coefficients, which gives a better quality reconstructed speech and also a more desirable frequency response than a one-tap pitch prediction filter. The pseudo-three-tap pitch filter with one degree of freedom is of particular interest as no extra bits are needed to code the pitch coefficients.
The second new approach is to perform time scaling/shifting on the original speech minimizing further the minimum mean square error and allowing a smoother and more accurate reconstruction of the pitch structure. The time scaling technique allows a saving of 1 bit in coding the pitch parameters while maintaining very closely the quality of the reconstructed speech. In addition, no extra bits are needed for the time scaling operation as no extra side information has to be transmitted to the receiver.
Yan, Ming. "VLSI architectures for speech and image coding applications". Thesis, Queen's University Belfast, 1989. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.356855.
Texto completoZemouri, Rachid. "Data compression of speech using sub-band coding". Thesis, University of Newcastle Upon Tyne, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.316094.
Texto completoDavis, Andrew J. "Waveform coding of speech and voiceband data signals". Thesis, University of Liverpool, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.232946.
Texto completoLAMARE, RODRIGO CAIADO DE. "SPEECH CODING AT AVERAGE RATES BELOW 2KB/S". PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2001. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=1873@1.
Texto completoEsta dissertação propõe algoritmos para codificações de voz a taxas médias em torno de 1,2 Kb/s. Um esquema de quantização vetorial preditiva chaveada com desempenho superior aos esquemas previamente descritos na literatura é proposto e avaliado em canal com ou sem ruído. Detectores eficientes de período fundamental e de sons oclusivos e fricativos são examinados e adaptados ao codificador proposto. Técnicas de exitação a baixas taxas de bits são investigadas a fim de reproduzir uma boa qualidade de voz decodificada. O modelo de exitação mista em multi-bandas com três sub-bandas é adotado para codificar os quadros sonoros. Para os quadros surdos são empregadas técnicas de modelagem e síntese de sinais fricativos e oclusivos, capazes de oferecer qualidade de voz satisfatória, reduzindo a taxa de bits destes quadros para apenas 0,4 Kb/s. Técnicas de pós-filtragem para reduzir o ruído de codificação e melhorar a qualidade de voz reconstruída são também examinadas e comparadas em uma mesma plataforma. Para reduzir o nível de ruído ambiente são ainda analisados métodos de supressão de ruído. Finalmente, o codificador proposto é comparado ao padrão norte-americano Mixed Excitation Linear Prediction (MELP), por meios de teste de comparação do tipo A/B. Os testes realizados indicam que o sistema proposto, operando a 1,2 Kb/s, apresenta qualidade de voz ligeiramente superior ao MELP, operando a 2,4 Kb/s. Para situações de transcodificação, o codificador proposto também apresenta desempenho superior ao MELP.
This dissertation presents algorithms to encode at an avarage bit rate of 1.2 Kb/s. A novel switched-predictive vector quantiser technique that outperforms previously reported schemes is proposed and assessed under noise-free and noisy channels. Efficient detectors for the pitch period and fricative and stop sounds are examined and adapted to the proposed coder. Low bit rate excitation methods are investigated in order to reproduce rather high quality speech. A mixed multiband excitation approach with three sub-bands is employed to encode voiced frames. For unvoiced frames, fricatives and stops modelling and synthesis techniques are used. This approach has shown to provide high quality synthesised speech, whilts it reduces the bit rate to only 0.4 Kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, post- filtering techniques are analysed and compared on the same plataform. To reduce background noise, noise suppression methods are also examined. Finally, the propose coder is evaluated against the North American Mixed Prediction (MELP) coder, through A/B comparison tests. Assessment results have shown that the proposed system, operating at 1.2 Kb/s, slightly outperformed the MELP coder, operating at 2.4 Kb/s. For tandem connection situations, the proposed algorithm has presented a superior performance than the MELP coder.
Esta disertación propone algoritmos para codificaciones de voz a tasas medias en torno de 1,2 Kb/s. Se propone un esquema de cuantización vectorial predictiva, con desempeño superior a los esquemas previamente descritos en la literatura. Este esquema se evalúa en canal con o sin ruido. Se examinan detectores eficientes de período fundamental y de sueños oclusivos y fricativos se adaptan al codificador propuesto. Técnicas de exitación a bajas tasas de bits son investigadas a fin de reproducir una boa calidad de voz decodificada. Se adopta el modelo de exitación mixta en multi-bandas con tres sub-bandas para codificar los cuadros sonoros. Para los cuadros surdos se emplean técnicas de modelación y síntesis de señales fricativos y oclusivos, capaces de ofrecer calidad de voz satisfactoria, reduciendo la tasa de bits de estos cuadros para apenas 0,4 Kb/s. También se examinan y se comparan las técnicas de pós-filtragen para reducir el ruido de codificación y mejorar la calidad de voz reconstruída. Para reducir el nível de ruído ambiente se analizan métodos de supresión de ruido. Finalmente, el codificador propuesto se compara al padrón norteamericano Mixed Excitation Lineal Prediction (MELP), por medio de pruebas de comparación del tipo LA/B. Las pruebas realizadas indican que el sistema propuesto, operando a 1,2 Kb/s, presenta calidad de voz ligeramente superior al MELP, operando a 2,4 Kb/s. Para situaciones de transcodificación, el codificador propuesto también presenta desempeño superior al MELP.
Savvides, Vasos E. "Perceptual models in speech quality assessment and coding". Thesis, Loughborough University, 1988. https://dspace.lboro.ac.uk/2134/36273.
Texto completoLeBlanc, Wilfrid P. (Wilfrid Paul) Carleton University Dissertation Engineering Electrical. "Speech coding at low to medium bit rates". Ottawa, 1992.
Buscar texto completoLucas, Adrian Edward. "Acoustic level speech recognition". Thesis, University of Surrey, 1991. http://epubs.surrey.ac.uk/2819/.
Texto completoCoetzee, H. J. "The development of a new objective speech quality measure for speech coding applications". Diss., Georgia Institute of Technology, 1990. http://hdl.handle.net/1853/15474.
Texto completoMurray, Alan. "An investigation into a speaker dependent coding system". Thesis, Leeds Beckett University, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.321413.
Texto completoMcCourt, Paul. "Transform vector quantisation of speech at low bit rates". Thesis, Queen's University Belfast, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.282252.
Texto completoKura, Vijay. "Novel Pitch Detection Algorithm With Application to Speech Coding". ScholarWorks@UNO, 2003. http://scholarworks.uno.edu/td/52.
Texto completoPeters, Richard Alan II. "A LINEAR PREDICTION CODING MODEL OF SPEECH (SYNTHESIS, LPC, COMPUTER, ELECTRONIC)". Thesis, The University of Arizona, 1985. http://hdl.handle.net/10150/291240.
Texto completoLee, Kwan Yee. "Analysis-by-synthesis linear predictive coding". Thesis, University of Surrey, 1990. http://epubs.surrey.ac.uk/844188/.
Texto completoMurray, Iain Robert. "Simulating emotion in synthetic speech". Thesis, University of Dundee, 1989. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.306550.
Texto completoNiranjan, Mahesan. "Modelling and classifying speech patterns". Thesis, University of Cambridge, 1990. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.303223.
Texto completoWu, Lizhong. "Speech processing with neural networks". Thesis, University of Cambridge, 1992. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.259529.
Texto completoTaft, Daniel Adam. "Cochlear implant sound coding with across-frequency delays". Connect to thesis, 2009. http://repository.unimelb.edu.au/10187/5783.
Texto completoBefore incorporating cochlear delays into a cochlear implant processor, a set of suitable delays was determined with a psychoacoustic calibration to pitch perception, since normal cochlear delays are a function of frequency. The first experiment assessed the perception of pitch evoked by electrical stimuli from cochlear implant electrodes. Six cochlear implant users with acoustic hearing in their non-implanted ears were recruited for this, since they were able to compare electric stimuli to acoustic tones. Traveling wave delays were then computed for each subject using the frequencies matched to their electrodes. These were similar across subjects, ranging over 0-6 milliseconds along the electrode array.
The next experiment applied the calibrated delays to the ACE strategy filter outputs before maxima selection. The effects upon speech perception in noise were assessed with cochlear implant users, and a small but significant improvement was observed. A subsequent sensitivity analysis indicated that accurate calibration of the delays might not be necessary after all; instead, a range of across-frequency delays might be similarly beneficial.
A computational investigation was performed next, where a corpus of recorded speech was passed through the ACE cochlear implant sound processing strategy in order to determine how across-frequency delays altered the patterns of stimulation. A range of delay vectors were used in combination with a number of processing parameter sets and noise levels. The results showed that additional stimuli from broadband sounds (such as the glottal pulses of vowels) are selected when frequency bands are desynchronized with across-frequency delays. Background noise contains fewer dominant impulses than a single talker and so is not enhanced in this way.
In the following experiment, speech perception with an ensemble of across-frequency delays was assessed with eight cochlear implant users. Reverse cochlear delays (high frequency delays) were equivalent to conventional cochlear delays. Benefit was diminished for larger delays. Speech recognition scores were at baseline with random delay assignments. An information transmission analysis of speech in quiet indicated that the discrimination of voiced cues was most improved with across-frequency delays. For some subjects, this was seen as improved vowel discrimination based on formant locations and improved transmission of the place of articulation of consonants.
A final study indicated that benefits to speech perception with across-frequency delays are diminished when the number of maxima selected per frame is increased above 8-out-of-22 frequency bands.