Conferences related to Speech Applications

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2023 Annual International Conference of the IEEE Engineering in Medicine & Biology Conference (EMBC)

The conference program will consist of plenary lectures, symposia, workshops and invitedsessions of the latest significant findings and developments in all the major fields of biomedical engineering.Submitted full papers will be peer reviewed. Accepted high quality papers will be presented in oral and poster sessions,will appear in the Conference Proceedings and will be indexed in PubMed/MEDLINE.


ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)

The ICASSP meeting is the world's largest and most comprehensive technical conference focused on signal processing and its applications. The conference will feature world-class speakers, tutorials, exhibits, and over 50 lecture and poster sessions.


IECON 2020 - 46th Annual Conference of the IEEE Industrial Electronics Society

IECON is focusing on industrial and manufacturing theory and applications of electronics, controls, communications, instrumentation and computational intelligence.


2019 20th International Conference on Solid-State Sensors, Actuators and Microsystems & Eurosensors XXXIII (TRANSDUCERS & EUROSENSORS XXXIII)

The world's premiere conference in MEMS sensors, actuators and integrated micro and nano systems welcomes you to attend this four-day event showcasing major technological, scientific and commercial breakthroughs in mechanical, optical, chemical and biological devices and systems using micro and nanotechnology.The major areas of activity in the development of Transducers solicited and expected at this conference include but are not limited to: Bio, Medical, Chemical, and Micro Total Analysis Systems Fabrication and Packaging Mechanical and Physical Sensors Materials and Characterization Design, Simulation and Theory Actuators Optical MEMS RF MEMS Nanotechnology Energy and Power


2018 15th IEEE Annual Consumer Communications & Networking Conference (CCNC)

IEEE CCNC 2018 will present the latest developments and technical solutions in the areas of home networking, consumer networking, enabling technologies (such as middleware) and novel applications and services. The conference will include a peer-reviewed program of technical sessions, special sessions, business application sessions, tutorials, and demonstration sessions


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Periodicals related to Speech Applications

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Audio, Speech, and Language Processing, IEEE Transactions on

Speech analysis, synthesis, coding speech recognition, speaker recognition, language modeling, speech production and perception, speech enhancement. In audio, transducers, room acoustics, active sound control, human audition, analysis/synthesis/coding of music, and consumer audio. (8) (IEEE Guide for Authors) The scope for the proposed transactions includes SPEECH PROCESSING - Transmission and storage of Speech signals; speech coding; speech enhancement and noise reduction; ...


Biomedical Engineering, IEEE Transactions on

Broad coverage of concepts and methods of the physical and engineering sciences applied in biology and medicine, ranging from formalized mathematical theory through experimental science and technological development to practical clinical applications.


Circuits and Systems I: Regular Papers, IEEE Transactions on

Part I will now contain regular papers focusing on all matters related to fundamental theory, applications, analog and digital signal processing. Part II will report on the latest significant results across all of these topic areas.


Communications Magazine, IEEE

IEEE Communications Magazine was the number three most-cited journal in telecommunications and the number eighteen cited journal in electrical and electronics engineering in 2004, according to the annual Journal Citation Report (2004 edition) published by the Institute for Scientific Information. Read more at http://www.ieee.org/products/citations.html. This magazine covers all areas of communications such as lightwave telecommunications, high-speed data communications, personal communications ...


Communications, IEEE Transactions on

Telephone, telegraphy, facsimile, and point-to-point television, by electromagnetic propagation, including radio; wire; aerial, underground, coaxial, and submarine cables; waveguides, communication satellites, and lasers; in marine, aeronautical, space and fixed station services; repeaters, radio relaying, signal storage, and regeneration; telecommunication error detection and correction; multiplexing and carrier techniques; communication switching systems; data communications; and communication theory. In addition to the above, ...


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Most published Xplore authors for Speech Applications

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Xplore Articles related to Speech Applications

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An approach to architectural enhancement for embedded speech applications

19th International Conference on VLSI Design held jointly with 5th International Conference on Embedded Systems Design (VLSID'06), 2006

Advances in human computer interaction (HCI) technology has resulted in widespread development of natural language and speech applications. These applications are known to be computationally different from other end-user applications. In this paper, the architectural features of text to speech (TTS) and automatic speech recognition (ASR) applications have been evaluated and cost effective optimal architectural solutions have been suggested. The ...


3.3 V mixed-mode IC design using switched-current techniques for speech applications

IEE Proceedings - Circuits, Devices and Systems, 1997

A low-cost, low-voltage, mixed-mode integrated circuit for speech applications is presented. The mixed-mode IC can directly preamplify speech signals with the aid of automatic gain control, convert amplified signals to 1-bit digital codes at 24 kbit/s, store the converted codes in memory, and replay the stored codes through an on-chip current-input voltage-output power amplifier. Switched-current techniques are used to implement ...


An application specific DSP for speech applications

IEEE Transactions on Consumer Electronics, 1993

A 16-bit digital signal processor core, designed for speech applications in telecommunications and consumer electronics is described. It enables low-cost, low-power DSP processing with several levels of modularity, permitting efficient DSP-based ASIC development. The DSP core can perform speech compression for applications such as digital answering machines and cellular phones.<<ETX>>


An improved critical-band transform processor for speech applications

2004 IEEE International Symposium on Circuits and Systems (IEEE Cat. No.04CH37512), 2004

This paper presents a computationally efficient algorithm of critical-band transform (CBT) for approximating the critical-band filtering of human ear. The CBT consists of a constant-bandwidth transform in the lower frequency range and a Brown constant-Q transform in the higher frequency range. A 21-band CBT at a sampling rate of 16 kHz is proposed with better approximation to the human critical-band ...


Dynamic Models for Speech Applications

Decision Making Under Uncertainty: Theory and Application, None

This chapter provides an overview of speech applications and the role of basic modeling and decision techniques in these applications. We will focus on the area of speech recognition involving extracting information from speech. Classically, the main piece of information extracted from a speech signal is the sequence of words. In this chapter, we will describe the basic speech recognition ...


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Educational Resources on Speech Applications

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IEEE-USA E-Books

  • An approach to architectural enhancement for embedded speech applications

    Advances in human computer interaction (HCI) technology has resulted in widespread development of natural language and speech applications. These applications are known to be computationally different from other end-user applications. In this paper, the architectural features of text to speech (TTS) and automatic speech recognition (ASR) applications have been evaluated and cost effective optimal architectural solutions have been suggested. The results have shown that low cost solutions such as ISA extension and feature addition to an existing embedded architecture can lead to appreciable improvement in performance for these applications. This work also presents a design space exploration to suggest an optimized VLIW architecture for TTS and ASR applications.

  • 3.3 V mixed-mode IC design using switched-current techniques for speech applications

    A low-cost, low-voltage, mixed-mode integrated circuit for speech applications is presented. The mixed-mode IC can directly preamplify speech signals with the aid of automatic gain control, convert amplified signals to 1-bit digital codes at 24 kbit/s, store the converted codes in memory, and replay the stored codes through an on-chip current-input voltage-output power amplifier. Switched-current techniques are used to implement a filter with a 3.4 kHz bandwidth and a 96 kHz sampling frequency, and also an adaptive delta modulator (ADM) with a 24 kHz sampling frequency. Measured dynamic range of the SI ADM combined with the SI filter is greater than 40 dB. This mixed-mode IC, operating with a 3.3 V supply voltage, was fabricated by a 1.5-/spl mu/m single-poly single-metal CMOS process. The whole system has been tested using human voice and results show that it is highly suitable for low-cost speech applications.

  • An application specific DSP for speech applications

    A 16-bit digital signal processor core, designed for speech applications in telecommunications and consumer electronics is described. It enables low-cost, low-power DSP processing with several levels of modularity, permitting efficient DSP-based ASIC development. The DSP core can perform speech compression for applications such as digital answering machines and cellular phones.<<ETX>>

  • An improved critical-band transform processor for speech applications

    This paper presents a computationally efficient algorithm of critical-band transform (CBT) for approximating the critical-band filtering of human ear. The CBT consists of a constant-bandwidth transform in the lower frequency range and a Brown constant-Q transform in the higher frequency range. A 21-band CBT at a sampling rate of 16 kHz is proposed with better approximation to the human critical-band analysis, and significantly fewer computations than other methods. Simulation results also verified its suitability for performing short-time spectral analysis on speech. A VLSI architecture for the hardware implementation of the 21-band CBT is proposed. In comparison to a 256-point in-place FFT processor, the proposed algorithm and architecture result in a 65.7% reduction in on-chip memory accesses, 56.9% reduction in real multiplications and 15.4% reduction in real additions. The proposed 21-band CBT processor is therefore an improved single-chip solution in low-power speech applications.

  • Dynamic Models for Speech Applications

    This chapter provides an overview of speech applications and the role of basic modeling and decision techniques in these applications. We will focus on the area of speech recognition involving extracting information from speech. Classically, the main piece of information extracted from a speech signal is the sequence of words. In this chapter, we will describe the basic speech recognition application as well as other concepts such as gender, language, and speaker recognition.

  • N/sup 2/LMS: enhanced LMS algorithm for speech applications

    A modified LMS algorithm, for the echo control of speech signals in a conference environment, has been analysed. Initial results show that the algorithm can provide lower misadjustment and more uniform speech-band cancellation compared with the conventional LMS algorithm at equal convergence rates, with very little additional computation.

  • Efficient VLSI Architecture of Lifting-Based Wavelet Packet Transform for Audio and Speech Applications

    This paper presents a novel VLSI architecture for discrete wavelet packet transform (DWPT). By exploiting the in-place nature of the DWPT algorithm, this architecture has an efficient pipeline structure to implement high- throughput processing. Folded architecture for lifting-based wavelet filters is proposed to compute wavelet butterflies in different groups simultaneously, at each decomposition level. Internal pipelining and by-pass mode are employed on each processing element to increase computation throughput and provide easy configuration for arbitrary decomposition, respectively. According to the comparison results, our proposed VLSI architecture is more efficient than previous proposed architectures in terms of arithmetic operations, storage requirement, and throughput

  • Speech applications created by a microcomputer: Discover the advantages of newly developed single-chip microcomputers for digital signal processing in speech applications

    The authors discuss the use of the TMS32010 general purpose digital signal processing microcomputer, which has the flexibility to be programmed as a speech synthesizer, an analyzer, or a recognizer. They cover vocoding, speech recognition, a text-to-speech system, and a general-purpose speech system.

  • NLPBench: A Tool for Studying the Architectural Characteristics of Natural Langauge and Speech Applications

    The study of architectures for natural language and speech applications has been necessitated by the large scale proliferation of such applications into desktop and embedded computers. This paper proposes NLPBench a benchmark for analyzing statistical natural language applications. This study validates the need for such a benchmark by demonstrating the lack of such applications in the currently existing benchmarks. A comparison between architectural characteristics of the applications in NLPBench is done with currently existing benchmarks to further prove the need for such a benchmark

  • Gradient encoding for Low-Bit-Rate stored speech applications

    In stored speech applications, the waveform of the message is completely specified and can be effectively used to reduce the bit rate at which the message may be synthesized. In gradient encoding, we propose to match the gradient of the output wave of the differential decoder with the required gradient between discrete clock cycles. When the required gradient is very steep the bit pattern selected maximizes the rate of change of the decoder voltage, otherwise appropriate bits of opposite polarity are inserted to match the amplitude of the decoder voltage with the required voltage at the discrete clock cycles. The performances of gradient encoding and conventional encoding are presented as corresponding signal-to-noise ratios under different inputs and circuit conditions. Further, our preliminary results indicate that gradient encoding can lead to comparable quality of speech at about half the bit rate of the conventional encoding between 32 to 24 kbaud.



Standards related to Speech Applications

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IEEE Standard Methods for Measuring Transmission Performance of Analog and Digital Telephone Sets, Handsets, and Headsets

This standard provides the techniques for objective measurement of electroacoustic characteristics of analog and digital telephones, handsets and headsets. Application is in the frequency range from 100 Hz to 8500 Hz. Although not specifically within the scope of this standard, the methods described are generally applicable to a wide variety of other communications equipment, including cordless, wireless and mobile communications ...