Thematische Bibliographien / Speech Communication. Engineering, Electronics and Electrical

Inhaltsverzeichnis

  1. Zeitschriftenartikel
  2. Dissertationen
  3. Bücher
  4. Buchteile
  5. Konferenzberichte

Auswahl der wissenschaftlichen Literatur zum Thema „Speech Communication. Engineering, Electronics and Electrical“

Autor: Grafiati
Veröffentlicht am 4. Juni 2021
Zuletzt aktualisiert am 11. Februar 2022

Geben Sie eine Quelle nach APA, MLA, Chicago, Harvard und anderen Zitierweisen an

Wählen Sie eine Art der Quelle aus:

Machen Sie sich mit den Listen der aktuellen Artikel, Bücher, Dissertationen, Berichten und anderer wissenschaftlichen Quellen zum Thema "Speech Communication. Engineering, Electronics and Electrical" bekannt.

Neben jedem Werk im Literaturverzeichnis ist die Option "Zur Bibliographie hinzufügen" verfügbar. Nutzen Sie sie, wird Ihre bibliographische Angabe des gewählten Werkes nach der nötigen Zitierweise (APA, MLA, Harvard, Chicago, Vancouver usw.) automatisch gestaltet.

Sie können auch den vollen Text der wissenschaftlichen Publikation im PDF-Format herunterladen und eine Online-Annotation der Arbeit lesen, wenn die relevanten Parameter in den Metadaten verfügbar sind.

Zeitschriftenartikel zum Thema "Speech Communication. Engineering, Electronics and Electrical"

1

IFUKUBE, TOORU. „Exciting Challenge of Biomedical Engineering. Auditory Disorder and Speech Communication.“ Journal of the Institute of Electrical Engineers of Japan 119, Nr. 11 (1999): 679–81. http://dx.doi.org/10.1541/ieejjournal.119.679.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
2

Newell, A. F. „Speech communication technology—lessons from the disabled“. Electronics and Power 32, Nr. 9 (1986): 661. http://dx.doi.org/10.1049/ep.1986.0389.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
3

Zhang, Yu, Ming Dai, Yiman Hua und Gonghuan Du. „Hyperchaotic synchronisation scheme for digital speech communication“. Electronics Letters 35, Nr. 24 (1999): 2087. http://dx.doi.org/10.1049/el:19991411.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
4

Moller, Sebastian, und Richard Heusdens. „Objective Estimation of Speech Quality for Communication Systems“. Proceedings of the IEEE 101, Nr. 9 (September 2013): 1955–67. http://dx.doi.org/10.1109/jproc.2013.2241374.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
5

Allen, J. „A perspective on man-machine communication by speech“. Proceedings of the IEEE 73, Nr. 11 (1985): 1541–50. http://dx.doi.org/10.1109/proc.1985.13339.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
6

Weng, Zhenzi, und Zhijin Qin. „Semantic Communication Systems for Speech Transmission“. IEEE Journal on Selected Areas in Communications 39, Nr. 8 (August 2021): 2434–44. http://dx.doi.org/10.1109/jsac.2021.3087240.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
7

Karjalainen, Matti. „Speech communication, human and machine“. Signal Processing 15, Nr. 2 (September 1988): 217–18. http://dx.doi.org/10.1016/0165-1684(88)90074-6.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
8

Rahmani, M., N. Yousefian und A. Akbari. „Energy-based speech enhancement technique for hands-free communication“. Electronics Letters 45, Nr. 1 (2009): 85. http://dx.doi.org/10.1049/el:20092177.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
9

Deng, Zongyuan, Xi Shao, Zhen Yang und Baoyu Zheng. „A novel covert speech communication system and its implementation“. Journal of Electronics (China) 25, Nr. 6 (November 2008): 737–45. http://dx.doi.org/10.1007/s11767-007-0099-8.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
10

Kawaguchi, Nobuo, Kazuya Takeda und Fumitada Itakura. „Multimedia Corpus of In-Car Speech Communication“. Journal of VLSI Signal Processing-Systems for Signal, Image, and Video Technology 36, Nr. 2/3 (Februar 2004): 153–59. http://dx.doi.org/10.1023/b:vlsi.0000015094.60008.dc.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
Mehr Quellen

Dissertationen zum Thema "Speech Communication. Engineering, Electronics and Electrical"

1

Othman, Noor Shamsiah. „Wireless speech and audio communications“. Thesis, University of Southampton, 2008. https://eprints.soton.ac.uk/64488/.

Der volle Inhalt der Quelle
Annotation:
The limited applicability of Shannon’s separation theorem in practical speech/audio systems motivates the employment of joint source and channel coding techniques. Thus, considerable efforts have been invested in designing various types of joint source and channel coding schemes. This thesis discusses two different types of Joint Source and Channel Coding (JSCC) schemes, namely Unequal Error Protection (UEP) aided turbo transceivers as well as Iterative Source and Channel Decoding (ISCD) exploiting the residual redundancy inherent in the source encoded parameters. More specifically, in Chapter 2, two different UEP JSCC philosophies were designed for wireless audio and speech transmissions, namely a turbo-detected UEP scheme using twin-class convolutional codes and another turbo detector using more sophisticated Irregular Convolutional Codes (IRCC). In our investigations, the MPEG-4 Advanced Audio Coding (AAC), the MPEG-4 Transform-Domain Weighted Interleaved Vector Quantization (TwinVQ) and the Adaptive MultiRate WideBand (AMR-WB) audio/speech codecs were incorporated in the sophisticated UEP turbo transceiver, which consisted of a threestage serially concatenated scheme constituted by Space-Time Trellis Coding (STTC), Trellis Coded Modulation (TCM) and two different-rate Non-Systematic Convolutional codes (NSCs) used for UEP. Explicitly, both the twin-class UEP turbo transceiver assisted MPEG-4 TwinVQ and the AMR-WB audio/speech schemes outperformed their corresponding single-class audio/speech benchmarkers by approximately 0.5 dB, in terms of the required Eb/N0, when communicating over uncorrelated Rayleigh fading channels. By contrast, when employing the MPEG-4 AAC audio codec and protecting the class-1 audio bits using a 2/3-rate NSC code, a more substantial Eb/N0 gain of about 2 dB was achieved. As a further design alternative, we also proposed a turbo transceiver employing IRCCs for the sake of providing UEP for the AMR-WB speech codec. The resultant UEP schemes exhibited a better performance when compared to the corresponding Equal Error Protection (EEP) benchmark schemes, since the former protected the audio/speech bits according to their sensitivity. The proposed UEP aided system using IRCCs exhibits an Eb/N0 gain of about 0.4 dB over the EEP system employing regular convolutional codes, when communicating over AWGN channels, at the point of tolerating a SegSNR degradation of 1 dB. In Chapter 3, a novel system that invokes jointly optimised ISCD for enhancing the error resilience of the AMR-WB speech codec was proposed and investigated. The resultant AMR-WB coded speech signal is protected by a Recursive Systematic onvolutional (RSC) code and transmitted using a non-coherently detected Multiple-Input Multiple-Output (MIMO) Differential Space-Time Spreading (DSTS) scheme. To further enhance the attainable system performance and to maximise the coding advantage of the proposed transmission scheme, the system is also combined with multi-dimensional Sphere Packing (SP) modulation. The AMR-WB speech decoder was further developed for the sake of accepting the a priori information passed to it from the channel decoder as extrinsic information, where the residual redundancy inherent in the AMR-WB encoded parameters was exploited. Moreover, the convergence behaviour of the proposed scheme was evaluated with the aid of both Three-Dimensional (3D) and Two-Dimenstional (2D) EXtrinsic Information Transfer (EXIT) charts. The proposed scheme benefitted from the exploitation of the residual redundancy inherent in the AMR-WB encoded parameters, where an approximately 0.5 dB Eb/N0 gain was achieved in comparison to its corresponding hard speech decoding based counterpart. At the point of tolerating a SegSNR degradation of 1 dB, the advocated scheme exhibited an Eb/N0 gain of about 1.0 dB in comparison to the benchmark scheme carrying out joint channel decoding and DSTS aided SP-demodulation in conjunction with separate AMR-WB decoding, when communicating over narrowband temporally correlated Rayleigh fading channels. In Chapter 4, two jointly optimized ISCD schemes invoking the soft-output AMRWB speech codec using DSTS assisted SP modulation were proposed. More specifically, the soft-bit assisted iterative AMR-WB decoder’s convergence characteristics were further enhanced by using Over-Complete source-Mapping (OCM), as well as a recursive precoder. EXIT charts were used to analyse the convergence behaviour of the proposed turbo transceivers using the soft-bit assisted AMR-WB decoder. Explicitly, the OCM aided AMR-WB MIMO transceiver exhibits an Eb/N0 gain of about 3.0 dB in comparison to the benchmark scheme also using ISCD as well as DSTS aided SP-demodulation, but dispensing with the OCM scheme, when communicating over narrowband temporally correlated Rayleigh fading channels. Finally, the precoded soft-bit AMR-WB MIMO transceiver exhibits an Eb/N0 gain of about 1.5 dB in comparison to the benchmark scheme dispensing with the precoder, when communicating over narrowband temporally correlated Rayleigh fading channels.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
2

Shen, Donglin. „Emulation study of speech communications over ATM networks“. Thesis, University of Ottawa (Canada), 1996. http://hdl.handle.net/10393/9544.

Der volle Inhalt der Quelle
Annotation:
Speech communications over ATM networks is one of the important issues in the broadband ISDN. Although CCITT Study Group XIII has already proposed the Draft Recommendation I.121 on speech communications over broadband ISDN, there are many open issued to be further studied before the effective deployment of speech communications over broadband ISDN can take place. In this thesis, several issues on speech transmission over ATM network have been studied, such as packetization delay, network queuing delay, digital speech encoding algorithms and PVR algorithm. A packetized speech emulation device is proposed to provide the capability of subjective speech transmission quality evaluation over ATM network or other kind of packetized network. The boundary of speech transmission quality degradation that human hearing can tolerate against information loss rate, delay fluctuation and encoding mechanism are found through emulation. The echo effect in ATM network, which is a primary issue in speech communications is also discussed in the thesis. In particular, the emphasis is given to the specification, design and implementation of ATM speech emulator which consists of two subsystems: ATM Network Simulator (ATMNS) and Speech Transmission Emulator (STE). ATMNS has been specified according to the results of ATM network performance analysis, and STE is based on ATM specifications recommended by CCITT. A prototype of the emulator has been implemented on a personal computer and DSP5600 development system with a special designed audio interface to interconnect phone sets to the DSP5600 AD input. The software had been written in "C" and DSP assembly language. Subjective evaluation are conducted in terms of the following factors: Cell discarding rate, different network queuing delay and fluctuation to different PVR algorithms. These factors are basic issues which may affect ATM speech transmission quality in an ATM network. Finally, test results under different network conditions are given.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
3

Rouchy, Christophe. „Systematic Design of Space-Time Convolutional Codes“. Thesis, University of California, Santa Cruz, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=1554232.

Der volle Inhalt der Quelle
Annotation:

Space-time convolutional code (STCC) is a technique that combines transmit diversity and coding to improve reliability in wireless fading channels. In this proposal, we demonstrate a systematic design of multi-level quadrature amplitude modulation (M-QAM) STCCs utilizing quadrature phase shift keying (QPSK) STCC as component codes for any number of transmit antennas. Morever, a low complexity decoding algorithm is introduced, where the decoding complexity increases linearly by the number of transmit antennas. The approach is based on utilizing a group interference cancellation technique also known as combined array processing (CAP) technique.

Finally, our research topic will explore: with the current approach, a scalable STTC with better performance as compared to space- time block code (STBC) combined with multiple trellis coded modulation (MTCM) also known as STBC-MTCM; the design of low complexity decoder for STTC; the combination of our approach with multiple-input multiple-output orthogonal frequency division multiplexing (MIMO-OFDM).

APA, Harvard, Vancouver, ISO und andere Zitierweisen
4

Ho, Wen Tsern 1977. „Clock and data recovery circuitry for high speed communication systems“. Thesis, McGill University, 2004. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=82494.

Der volle Inhalt der Quelle
Annotation:
The maturing of the telecommunications industry has seen the development and implementation of devices that work at high frequencies of the electromagnetic spectrum. With the rapid deployment of optical networks, there is an increasing demand for low-cost and efficient communications circuitry. In order to interface with such high frequency signals at lower cost, there has been a recent push for very high frequency circuits using low-cost fabrication technologies like digital CMOS.
This thesis investigates the usage of legacy architectures and the implementation of different topologies using digital CMOS technology. Various Clock and Data Recovery Phase-Locked Loops have been implemented using a 0.18mum CMOS technology, and the process from modeling to actual implementation will be presented. The design of the components of the loop, layout issues, and the performance of the various designs will be discussed. New fully-differential CMOS designs that are optimized for high-speed operation, yet providing stable lock with minimal jitter, with a targeted operation range from 1 GHz to 7 GHz, will be described in detail, as well as their operation and optimization.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
5

Tian, Xizhen. „Investigation of HBT preamplification for high speed optical communication systems“. Thesis, University of Ottawa (Canada), 2002. http://hdl.handle.net/10393/6273.

Der volle Inhalt der Quelle
Annotation:
A noise analysis for a Common-Collector-Cascode traveling wave HBT preamplifier is developed, resulting in an expression for the preamplifier's equivalent input noise current density. A photoreceiver, consisting of a P-I-N and GaAs HBT MMIC distributed amplifier, was implemented using Nortel's GaAs HBT (f T = 70GHz) process. The noise performance of the P-I-N preamplifier was predicted based on the noise analysis equations. The P-I-N preamplifier, having a measured bandwidth of 22GHz, displayed a measured average equivalent input noise current density of 24 pA/Hz . Good agreement was obtained between the predicted and measured noise performance. The analysis gives useful insight into the dominant noise contributions of the preamplifier. An 8-stage HBT distributed amplifier was successfully developed. By considering the various issues involved in its design, a design procedure for monolithic distributed amplifiers is presented. The implementation of the HBT preamplifier is described and its measured results are given. From the excellent agreement between the predicted and measured performance, the design method is considered validated. The successful operation of the distributed amplifier, which provides 15dB gain and 35GHz 3dB bandwidth, fulfills the objective of experimental verification. The implemented photoreceiver is the first to have a P-I-N mounted on the MMIC chip.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
6

Fan, Yongquan. „Accelerating jitter and BER qualifications of high speed serial communication interfaces“. Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86531.

Der volle Inhalt der Quelle
Annotation:
High-Speed Serial Interface (HSSI) devices have witnessed an increased use in communications. As a measure of how often bit errors happen, Bit Error Rate (BER) performance is of paramount importance in any communication interface. The bit errors in HSSIs are in large part due to jitter. This thesis investigates the topic of accelerating the jitter and BER testing and characterization [1].
The thesis first proposes a new algorithm, suitable for extrapolating the receiver jitter tolerance performance from higher BER regions down to the 10-12 level or lower [2]. This algorithm enables us to perform the jitter tolerance characterization and production test more than 1000 times faster [3]. Then an under-sampling based transmitter test scheme is presented. The scheme can accurately extract the transmitter jitter and finish the whole transmitter test within 100ms [4] while the test usually takes seconds. All the receiver and transmitter testing schemes have been successfully used on Automatic Test Equipment (ATE) to qualify millions of HSSIs with speed up to 6 Gigabits per second (Gbps).
The thesis also presents an external loopback-based testing scheme, where a novel jitter injection technique is proposed using the state-of-the-art phase delay lines. The scheme can be applied to test HSSIs with data rate up to 12.5 Gbps. It is also suitable for multi-lane HSSI testing with a lower cost than pure ATE solutions. By using high-speed relays, we combine the proposed ATE based approaches and the loopback approach along with an FPGA-based BER tester to provide a more versatile scheme for HSSI post-silicon validation, testing and debugging [5]. In addition, we further explore the unparallel advantages of our digital Gaussian noise generator in low BER evaluation [6].
Les interfaces sérielles à haute vitesse (interfaces HSSI) ont connu une utilisation accrue dans les télécommunications. Le taux d'erreur sur les bits (BER), mesure de la fréquence des erreurs, est d'une importance cruciale dans les interfaces modernes de télécommunication. Cette thèse traite de l'accélération de la caractérisation du vacillement et des tests BER.
Cette thèse propose tout d'abord un nouvel algorithme, approprié pour l'extrapolation de la performance de la tolérance au vacillement d'un récepteur pour un taux d'erreur sur les bits (BER) à un niveau de 10-12 ou moins. Cet algorithme permet de caractériser la tolérance au vacillement dans les tests de production plus de 1000 fois plus rapidement. Ensuite, une conception de transmetteur à sous-échantillonnage est présenté. Cette conception permet d'extraire précisément le vacillement du transmetteur et de compléter les tests de ce dernier en moins de 100 ms alors que ces tests durent normalement plusieurs secondes. Toutes les méthodes de test de récepteurs et de transmetteurs ont été utilisées avec succès sur un équipement d'éssai automatique (ATE) pour qualifier des millions d'interfaces HSSI à des vitesses allant jusqu'à 6 gigabits par seconde (6 Gbps).
Cette thèse présente aussi une conception de test en bouclage où une nouvelle méthode d'injection de vacillement est proposée en utilisant des lignes de délai de phase. Cette méthode peut être appliquée pour tester des interfaces HSSI avec un taux de transfer allant jusqu'à 12.5 Gbps. Elle permet aussi de tester des interface HSSI multi-lignes à un coût moindre qu'une solution utilisant un ATE. En utilisant des relais à haute vitesse, les approches sur ATE et par test en bouclage peuvent être combinées en incorporant un testeur de BER sur circuit intégré prédiffusé programmable (FPGA), ce qui permet une méthode de tests HSSI polyvalente pour la validation post-fabrication, les tests et le débogage. Finalement, nous explorons les avantages de notre générateur de bruit Gaussien dans l'évaluation de BER à bas niveau.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
7

Elsherif, Mohamed Asaad. „Mapping multiplexing technique (MMT) : a novel intensity modulated transmission format for high-speed optical communication systems“. Thesis, University of Nottingham, 2016. http://eprints.nottingham.ac.uk/33413/.

Der volle Inhalt der Quelle
Annotation:
There is a huge rapid growth in the deployment of data centers, mainly driven from the increasing demand of internet services as video streaming, e-commerce, Internet Of Things (IOT), social media, and cloud computing. This led data centers to experience an expeditious increase in the amount of network traffic that they have to sustain due to requirement of scaling with the processing speed of Complementary metal–oxide–semiconductor (CMOS) technology. On the other side, as more and more data centers and processing cores are on demand, as the power consumption is becoming a challenging issue. Unless novel power efficient methodologies are innovated, the information technology industry will be more liable to a future power crunch. As such, low complex novel transmission formats featuring both power efficiency and low cost are considered the major characteristics enabling large-scale, high performance data transmission environment for short-haul optical interconnects and metropolitan range data networks. In this thesis, a novel high-speed Intensity-Modulated Direct-Detection (IM/DD) transmission format named “Mapping Multiplexing Technique (MMT)” for high-speed optical fiber networks, is proposed and presented. Conceptually, MMT design challenges the high power consumption issue that exists in high-speed short and medium range networks. The proposed novel scheme provides low complex means for increasing the power efficiency of optical transceivers at an impactful tradeoff between power efficiency, spectral efficiency, and cost. The novel scheme has been registered as a patent (Malaysia PI2012700631) that can be employed for applications related but not limited to, short-haul optical interconnects in data centers and Metropolitan Area networks (MAN). A comprehensive mathematical model for N-channel MMT modulation format has been developed. In addition, a signal space model for the N-channel MMT has been presented to serve as a platform for comparison with other transmission formats under optical channel constraints. Especially, comparison with M-PAM, as meanwhile are of practical interest to expand the capacity for optical interconnects deployment which has been recently standardized for Ethernet IEEE 802.3bs 100Gb/s and in today ongoing investigation activities by IEEE 802.3 400Gb/s Ethernet Task Force. Performance metrics have been considered by the derivation of the average electrical and optical power for N-channel MMT symbols in comparison with Pulse Amplitude Modulation (M-PAM) format with respect to the information capacity. Asymptotic power efficiency evaluation in multi-dimensional signal space has been considered. For information capacity of 2, 3 and 4 bits/symbol, 2-channel, 3-channel and 4-channel MMT modulation formats can reduce the power penalty by 1.76 dB, 2.2 dB and 4 dB compared with 4-PAM, 8-PAM and 16-PAM, respectively. This enhancement is equivalent to 53%, 60% and 71% energy per bit reduction to the transmission of 2, 3 and 4 bits per symbol employing 2-, 3- and 4-channel MMT compared with 4-, 8- and 16-PAM format, respectively. One of the major dependable parameters that affect the immunity of a modulation format to fiber non-linearities, is the system baud rate. The propagation of pulses in fiber with bitrates in the order > 10G, is not only limited by the linear fiber impairments, however, it has strong proportionality with fiber intra-channel non-linearities (Self Phase Modulation (SPM), Intra-channel Cross-Phase Modulation (IXPM) and Intra-channel Four-Wave Mixing (IFWM)). Hence, in addition to the potential application of MMT in short-haul networks, the thesis validates the practicality of implementing N-channel MMT system accompanied by dispersion compensation methodologies to extend the reach of error free transmission (BER ≤ 10-12) for Metro-networks. N-Channel MMT has been validated by real environment simulation results to outperform the performance of M-PAM in tolerating fiber non-linearities. By the employment of pre-post compensation to tolerate both residual chromatic dispersion and non-linearity, performance above the error free transmission limit at 40Gb/s bit rate have been attained for 2-, 3- and 4-channel MMT over spans lengths of up to 1200Km, 320 Km and 320 Km, respectively. While, at an aggregated bit rate of 100 Gb/s, error free transmission can be achieved for 2-, 3- and 4-channel MMT over spans lengths of up to 480 Km, 80 Km and 160 Km, respectively. At the same spectral efficiency, 4-channel MMT has realized a single channel maximum error free transmission over span lengths up to 320 Km and 160 Km at 40Gb/s and 100Gb/s, respectively, in contrast with 4-PAM attaining 240 Km and 80 Km at 40Gb/s and 100Gb/s, respectively.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
8

Leong, 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.

Der volle Inhalt der Quelle
Annotation:
In recent years, research in narrow-band digital speech coding has achieved good quality speech coders at low rates of 4.8 to 8.0 kb/s. This thesis examines the method proposed by W. B. Kleijn called prototype waveform interpolation (PWI) for coding the voiced sections of speech efficiently to achieve a coder below 4.8 kb/s while maintaining, even improving, the perceptual quality of current coders.
In 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.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
9

Abboud, Karim. „Wideband CELP speech coding“. Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56805.

Der volle Inhalt der Quelle
Annotation:
The purpose of this thesis is to study the coding of wideband speech and to improve on previous Code-Excited Linear Prediction (CELP) coders in terms of speech quality and bit rate. To accomplish this task, improved coding techniques are introduced and the operating bit rate is reduced while maintaining and even enhancing the speech quality.
the 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.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
10

Nour-Eldin, Amr. „Quantifying and exploiting speech memory for the improvement of narrowband speech bandwidth extension“. Thesis, McGill University, 2014. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=121195.

Der volle Inhalt der Quelle
Annotation:
Since its standardization in the 1960s, the bandwidth of traditional telephony speech has been limited to the 0.3–3.4 kHz narrowband range. Wideband speech reconstruction through artificial bandwidth extension (BWE) attempts to regenerate the highband frequency content above 3.4 kHz in the receiving end, thereby providing backward compatibility with existing networks. BWE schemes have primarily relied on memoryless mapping to capture the correlation between narrowband and highband spectra. In this thesis, we investigate exploiting speech memory—in reference to the long-term information in segments longer than the conventional 10–30ms frames—for the purpose of improving the cross-band correlation central to BWE. With speech durations of up to 600ms modelled through delta features, we first quantify the correlation between long-term parameterizations of the narrow and high frequency bands using information-theoretic measures in combination with statistical modelling based on Gaussian mixture models (GMMs) and vector quantization. In addition to showing thatthe inclusion of memory can indeed increase certainty about highband spectral content in joint-band GMMs by over 100%, our information-theoretic investigation also demonstrates that the gains achievable by such acoustic-only memory inclusion saturate at, roughly, the syllabic duration of 200ms. To translate the highband certainty gains achievable by memory inclusion into tangible BWE performance improvements, we subsequently propose two distinct and novel approaches for memory-inclusive GMM-based BWE where highband spectra are reconstructed given narrowband input by minimum mean-square error estimation. In the first approach, we incorporate delta features into the feature vector representations whose underlying cross-band correlations are to be modelled by joint-band GMMs. Due to their non-invertibility, however, the inclusion of delta features into the parameterization frontendin lieu of some of the conventional static features imposes a time-frequency information tradeoff. Accordingly, we propose an empirical optimization process to determine the optimal allocation of available dimensionalities among static and delta features such that the certainty about static highband content is maximized. Integrating frontend-based memory inclusion optimized as such into our memoryless BWE baseline system results in performance improvements that, while modest, involve no increases in extension-stage computational cost nor in training data requirements, thereby providing an easy and convenient means for exploiting speech dynamics to improve BWE performance. In our second approach, we focus on modelling the high-dimensional distributions underlying sequences of joint-band feature vectors. To that end, we extend the GMM framework by presenting a novel training approach where sequences of past frames are progressively used to estimate the parameters of high-dimensional temporally-extended GMMs in a tree-like time-frequency-localized fashion. The proposed approach thus breaks down the infeasible task of modelling high-dimensional distributions into a series of localized modelling operations with considerably lower complexity and fewer degrees of freedom. The proposed temporal-based GMM extension approach is presented in a manner that emphasizes its wide applicability to the general contexts of source-target conversion and high-dimensional modelling. By integrating temporally-extended GMMs into our memoryless BWE baseline system, we show that our model-based memory-inclusive BWE technique can outperform not only our first frontend-based approach, but also other comparable and oft-cited model-based techniques in the literature. Although this superior BWE performance is achieved at a significant increase in extension-stage computational costs, we nevertheless show these costs to be within the typical capabilities of modern communication devices such as tablets and smart phones.
Depuis sa normalisation dans les années 1960, la bande passante traditionnelle de la téléphonie de la parole a été limitée à la bande étroite de 0,3 à 3,4 kHz. La reconstruction de la parole à large bande à travers l'extension artificielle de la bande passante (EBP) essaye de régénérer la bande passante à haute fréquence au-dessus de 3,4 kHz au niveau du récepteur, ce qui permet la rétrocompatibilité avec les réseaux existants. Les travaux précédentes sur l'EBP ont principalement utilisé une cartographie sans mémoire pour modéliser la corrélation entre les spectres à bande étroite et ceux à haute fréquence. Dans cette thèse, nous étudions l'exploitation de la mémoire vocale en référence à l'information à long terme dans des segments plus longs que les cadres conventionnels de 10–30 ms; ceci est dans le but d'améliorer la corrélation inter-bande capitale pour l'EBP. Focalisant sur des durées de parole modélisées jusqu'à 600 ms par des coefficients delta, nous quantifions d'abord la corrélation entre les paramétrisations à long terme des bandes à bases et hautes fréquences en utilisant la théorie de l'information et la modélisation statistique basée sur des modèles de mélanges Gaussiens (GMMs) ainsi que la quantification vectorielle. En plus de montrer que l'inclusion de la mémoire peut en effet augmenter la certitude sur le contenu spectral de la haute bande dans des GMMs de bandes jointes de plus de 100%, notre étude démontre également que les gains réalisables par une telle inclusion sature, à peu près, à la durée syllabique de 200 ms. Afin de transformer ces gains théoriques de certitude sur la bande haute à des améliorations tangibles en performance de l'EBP, nous proposons ensuite deux nouvelles approches pour l'EBP avec mémoire qui sont basées sur des GMMs et où les spectres à haute bande sont reconstruits, sachant ceux de la bande étroite, par l'estimation de l'erreur quadratique moyenne. Dans la première approche, nous incorporons des coefficients delta dans les représentations vectorielles modélisées par des GMMs de bandes jointes. En raison de la non-inversibilité des coefficients delta, cependant, nous proposons un processus d'optimisation empirique pour déterminer l'allocation optimale des dimensionnalités disponibles parmi les paramètres statiques et coefficients delta de sorte que la certitude sur le contenu statique de la haute bande est maximisée. L'intégration de la mémoire optimisé de cette manière dans la paramétrisation de notre système de base d'EBP entraîne des améliorations de performances qui, bien que modestes, offrent un moyen facile et pratique pour exploiter les caractéristiques dynamiques de la parole afin d'améliorer les performances d'EBP. Dans notre deuxième approche, nous nous concentrons sur la modélisation des distributions de dimensionnalités élevées qui sous-tendent des séquences de vecteurs de paramètres de bandes conjointes. À cette fin, nous étendons le cadre de GMMs en présentant une nouvelle approche d'apprentissage où les séquences des cadres passés sont progressivement utilisées afin d'estimer les paramètres des GMMs de dimensionnalités élevées qui sont temporellement étendus d'une manière arborescente et localisée en temps-fréquence. En intégrant des GMMs temporellement étendus dans notre système de base d'EBP sans mémoire, nous montrons que cette technique d'EBP avec mémoire modelisée peut surpasser non seulement notre première approche basée sur les coefficients delta, mais aussi d'autres techniques souvent citées dans la littérature. Bien que cette performance supérieure est réalisée au coût d'une augmentation significative des calculs associés à l'étape d'extension, nous démontrons néanmoins que ces coûts sont conformes aux capacités typiques des appareils de communication modernes tels que les tablettes et les téléphones intelligents.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
Mehr Quellen

Bücher zum Thema "Speech Communication. Engineering, Electronics and Electrical"

1

service), SpringerLink (Online, Hrsg. VLSI for Wireless Communication. 2. Aufl. Boston, MA: Springer Science+Business Media, LLC, 2011.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
2

Deller, John R. Discrete-time processing of speech signals. New York: Institute of Electrical and Electronics Engineers, 2000.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
3

G, Proakis John, und Hansen John H. L, Hrsg. Discrete-time processing of speech signals. New York: Macmillan Pub. Co., 1993.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
4

Pathak, Manas A. Privacy-Preserving Machine Learning for Speech Processing. New York, NY: Springer New York, 2013.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
5

Topical, Meeting on Silicon Monolithic Integrated Circuits in RF Systems (1st 1998 Ann Arbor Michigan). 1998 Topical Meeting on Silicon Monolithic Integrated Circuits in RF Systems: Digest of papers : [17-18 September, 1998, Ann Arbor, Michigan, USA]. Piscataway, New Jersey: IEEE, 1998.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
6

Cheng, K. W. Eric. Electrical engineering writing handbook. Hong Kong: Hong Kong Polytechnic University, 2002.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
7

ICASSP (24th 1999 Phoenix, Ariz.). 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing: Proceedings : ICASSP99 Phoenix : March 15-19, 1999, Civic Plaza, Hyatt Regency, Phoenix, Arizona, U.S.A. Piscataway, NJ: IEEE, 1999.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
8

IEEE Radio Frequency Integrated Circuits Symposium (1998 Baltimore, Md.). 1998 IEEE Radio Frequency Integrated Circuits (RFIC) Symposium: Digest of papers. Herausgegeben von Mondal Jyothi, IEEE Microwave Theory and Techniques Society. und IEEE Electron Devices Society. New York: IEEE, 1998.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
9

1959-, Steyaert Michiel, und Sansen Willy M. C, Hrsg. Design of multi-bit delta-sigma A/D converters. Boston: Kluwer Academic Publishers, 2002.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
10

Mediterranean Electrotechnical Conference (6th 1991 Ljublana, Yugoslavia). 6th Mediterranean Electrotehnical [sic] Conference: Proceedings : 22-24 May 1991, "Cankarjev dom"--Cultural and Congress Center Ljubljana, Slovenia, Yugoslavia. New York: The Institute of Electrical and Electronics Engineers, 1991.

Den vollen Inhalt der Quelle finden
APA, Harvard, Vancouver, ISO und andere Zitierweisen
Mehr Quellen

Buchteile zum Thema "Speech Communication. Engineering, Electronics and Electrical"

1

Cheng, Li, Ning Yang, Ming Yan und Xiaodi Zhou. „Two Switched-Current Memory Cells for High-Speed Digital Communication System“. In Advanced Electrical and Electronics Engineering, 519–24. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-19712-3_66.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
2

Liu, Yu. „Research on Integration of Heterogeneous Wireless Access Communication Networks“. In Advanced Electrical and Electronics Engineering, 635–42. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-19712-3_81.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
3

Yang, Guojian, und Zhuo Wang. „Research on the Radiation Field of Mono-radiation Wide-Band Leaky Coaxial Cable Using for 3G Mobile Communication“. In Advanced Electrical and Electronics Engineering, 643–50. Berlin, Heidelberg: Springer Berlin Heidelberg, 2011. http://dx.doi.org/10.1007/978-3-642-19712-3_82.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
4

Brumberg, Jonathan S., Frank H. Guenther und Philip R. Kennedy. „An Auditory Output Brain–Computer Interface for Speech Communication“. In SpringerBriefs in Electrical and Computer Engineering, 7–14. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-36083-1_2.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
5

Walrand, Jean. „Speech Recognition: A“. In Probability in Electrical Engineering and Computer Science, 205–15. Cham: Springer International Publishing, 2021. http://dx.doi.org/10.1007/978-3-030-49995-2_11.

Der volle Inhalt der Quelle
Annotation:
AbstractSpeech recognition can be formulated as the problem of guessing a sequence of words that produces a sequence of sounds. The human brain is remarkably good at solving this problem, even though the same words correspond to many different sounds, because of accents or characteristics of the voice. Moreover, the environment is always noisy, to that the listeners hear a corrupted version of the speech.Computers are getting much better at speech recognition and voice command systems are now common for smartphones (Siri), automobiles (GPS, music, and climate control), call centers, and dictation systems. In this chapter, we explain the main ideas behind the algorithms for speech recognition and for related applications.The starting point is a model of the random sequence (e.g., words) to be recognized and of how this sequence is related to the observation (e.g., voice). The main model is called a hidden Markov chain. The idea is that the successive parts of speech form a Markov chain and that each word maps randomly to some sounds. The same model is used to decode strings of symbols in communication systems.Section 11.1 is a general discussion of learning. The hidden Markov chain model used in speech recognition and in error decoding is introduced in Sect. 11.2. That section explains the Viterbi algorithm. Section 11.3 discusses expectation maximization and clustering algorithms. Section 11.4 covers learning for hidden Markov chains.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
6

Li, Xiaoning, Xiaofeng Li, Zhiyong Zhang und Zhuo Zhang. „Automatic Speech Embedded Word Method“. In Proceedings of the 2012 International Conference on Communication, Electronics and Automation Engineering, 93–100. Berlin, Heidelberg: Springer Berlin Heidelberg, 2013. http://dx.doi.org/10.1007/978-3-642-31698-2_14.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
7

Devi, Jutika, und Pranayee Datta. „Nanoelectronics“. In Handbook of Research on 5G Networks and Advancements in Computing, Electronics, and Electrical Engineering, 20–35. IGI Global, 2021. http://dx.doi.org/10.4018/978-1-7998-6992-4.ch002.

Der volle Inhalt der Quelle
Annotation:
The passive circuit elements resistor, inductor, and capacitor, which are the basic building blocks of an electronic circuit, need to be scaled down for application in fifth-generation wireless communication networks. Due to the growing demands in memory and computational capacities of integrated circuits along with high processing and transmission data speed for next-generation, microelectronics will be replaced by nanoelectronics in the future. The concept of nanoscale network on chip system is expected to play an important role in the field of communication systems for designing new devices of ultra-high speed for long and short-range communication links, power efficient computing devices, high-density memory and logic, and ultrafast interconnects. This chapter focuses on the mechanism of tailoring, patterning, and manipulating optical signals using nanometer-scale structures that may play the role of lumped nanocircuit elements at optical domain when selected properly with tremendous promise for application for fifth-generation communication systems.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
8

Burks, Arthur W. „An Early Graduate Program in Computers and Communications“. In Perspectives on Adaptation in Natural and Artificial Systems. Oxford University Press, 2005. http://dx.doi.org/10.1093/oso/9780195162929.003.0010.

Der volle Inhalt der Quelle
Annotation:
This is the story of how, in 1957, John Holland, a graduate student in mathematics; Gordon Peterson, a professor of speech; the present writer, a professor of philosophy; and several other Michigan faculty started a graduate program in Computers and Communications—with John our first Ph.D. and, I believe, the world's first doctorate in this now-burgeoning field. This program was to become the Department of Computer and Communication Sciences in the College of Literature, Science, and the Arts about ten years later. It had arisen also from a research group at Michigan on logic and computers that I had established in 1949 at the request of the Burroughs Adding Machine Company. When I first met John in 1956, he was a graduate of MIT in electrical engineering, and one of the few people in the world who had worked with the relatively new electronic computers. He had used the Whirlwind I computer at MIT [33], which was a process-control variant of the Institute for Advanced Study (IAS) Computer [27]. He had also studied the 1946 Moore School Lectures on the design of electronic computers, edited by George Patterson [58]. He had then gone to IBM and helped program its first electronic computer, the IBM 701, the first commercial version of the IAS Computer. While a graduate student in mathematics at Michigan, John was also doing military work at the Willow Run Research Laboratories to support himself. And 1 had been invited to the Laboratories by a former student of mine, Dr. Jesse Wright, to consult with a small research group of which John was a member. It was this meeting that led to the University's graduate program and then the College's full-fledged department. The Logic of Computers Group, out of which this program arose, in part, then continued with John as co-director, though each of us did his own research. This anomaly of a teacher of philosophy meeting an accomplished electrical engineer in the new and very small field of electronic computers needs some explanation, one to be found in the story of the invention of the programmable electronic computer. For the first three programmable electronic computers (the manually programmed ENIAC and the automatically programmed EDVAC and Institute for Advanced Study Computer) and their successors constituted both the instrumentation and the subject matter of our new Graduate Program in Computers and Communications.
APA, Harvard, Vancouver, ISO und andere Zitierweisen
9

„Target speech detection using Gaussian mixture modeling of frequency bandwise power ratio for GSC-based beamforming“. In Electronics and Electrical Engineering, 203–6. CRC Press, 2015. http://dx.doi.org/10.1201/b18443-38.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
10

H√•kansson, Lars, Sven Johansson, Mattias Dahl, Per Sj√∂sten und Ingvar Claesson. „Noise Canceling Headsets for Speech Communication“. In Electrical Engineering & Applied Signal Processing Series. CRC Press, 2002. http://dx.doi.org/10.1201/9781420041262.ch12.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen

Konferenzberichte zum Thema "Speech Communication. Engineering, Electronics and Electrical"

1

O'Shaughnessy, Douglas. „Automatic speech recognition“. In 2015 Chilean Conference on Electrical, Electronics Engineering, Information and Communication Technologies (CHILECON). IEEE, 2015. http://dx.doi.org/10.1109/chilecon.2015.7400411.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
2

Pasha, Nadeem, und Roopa S. „Continuous Kannada Noisy Speech Recognition“. In 2018 International Conference on Recent Innovations in Electrical, Electronics & Communication Engineering (ICRIEECE). IEEE, 2018. http://dx.doi.org/10.1109/icrieece44171.2018.9009108.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
3

Ahmed, Safayet, Rafiqul Islam, Md Saniat Rahman Zishan, Mohammed Rabiul Hasan und Md Nahian Islam. „Electronic speaking system for speech impaired people: Speak up“. In 2015 International Conference on Electrical Engineering and Information Communication Technology (ICEEICT). IEEE, 2015. http://dx.doi.org/10.1109/iceeict.2015.7307401.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
4

Paul, Yogesh, Ram Avtar Jaswal und Sanjay Kajal. „Classification of EEG Based Imagine Speech Using Time Domain Features“. In 2018 International Conference on Recent Innovations in Electrical, Electronics & Communication Engineering (ICRIEECE). IEEE, 2018. http://dx.doi.org/10.1109/icrieece44171.2018.9008572.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
5

Singh, Moirangthem Tiken, Partha Pratim Barman und Rupjyoti Gogoi. „Speech Recognition Model for Assamese Language Using Deep Neural Network“. In 2018 International Conference on Recent Innovations in Electrical, Electronics & Communication Engineering (ICRIEECE). IEEE, 2018. http://dx.doi.org/10.1109/icrieece44171.2018.9008668.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
6

Habib, Zeeshan, Jan Sher Khan, Jawad Ahmad, Muazzam A. Khan und Fadia Ali Khan. „Secure speech communication algorithm via DCT and TD-ERCS chaotic map“. In 2017 4th International Conference on Electrical and Electronic Engineering (ICEEE). IEEE, 2017. http://dx.doi.org/10.1109/iceee2.2017.7935827.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
7

Boza-Quispe, Gustavo, Juan Montalvan-Figueroa, Jimmy Rosales-Huamani und Fabricio Puente-Mansilla. „A friendly speech user interface based on Google cloud platform to access a tourism semantic website“. In 2017 CHILEAN Conference on Electrical, Electronics Engineering, Information and Communication Technologies (CHILECON). IEEE, 2017. http://dx.doi.org/10.1109/chilecon.2017.8229578.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
8

Rama krishna, N. J., und A. Ravi shankar. „Experimental determination of Transient speed for an Electrical Machine“. In 2018 International Conference on Recent Innovations in Electrical, Electronics & Communication Engineering (ICRIEECE). IEEE, 2018. http://dx.doi.org/10.1109/icrieece44171.2018.9008979.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
9

Prem, Oshin, Bhavnesh Kumar und S. K. Jha. „Intelligent Speed Control of DC Servo Motor Drive“. In 2018 International Conference on Recent Innovations in Electrical, Electronics & Communication Engineering (ICRIEECE). IEEE, 2018. http://dx.doi.org/10.1109/icrieece44171.2018.9009138.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
10

Mora, Andres, Hernan Yagama, David Zorro und L. Fabian R. Jimenez. „Speed digital control for scale car via Bluetooth and Android“. In 2015 Chilean Conference on Electrical, Electronics Engineering, Information and Communication Technologies (CHILECON). IEEE, 2015. http://dx.doi.org/10.1109/chilecon.2015.7400364.

Der volle Inhalt der Quelle
APA, Harvard, Vancouver, ISO und andere Zitierweisen
Die Auswahlen zum Thema "Speech Communication. Engineering, Electronics and Electrical" für konkrete Quellenarten können Sie auch interessieren:
Zeitschriftenartikel Dissertationen Bücher
Wir bieten Rabatte auf alle Premium-Pläne für Autoren, deren Werke in thematische Literatursammlungen aufgenommen wurden. Kontaktieren Sie uns, um einen einzigartigen Promo-Code zu erhalten!

Zur Bibliographie