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voice coding experiences for developers with physical impairments

JOURNAL OF ENABLING TECHNOLOGIES 2024年第4期18卷 265-275页

作者： Creed, Chris Sarcar, Sayan Birmingham City Univ Human Comp Interact HCI Birmingham England Birmingham City Univ Fac Comp Engn & Built Environm Birmingham England

PurposeThis research explores the current working practices of voice coders with physical impairments and investigates their perceptions of a research prototype utilising a fixed grammar speech coding ***/methodology/approachSemi-structured interviews were initially conducted with seven voice coders with physical impairments to understand their development practices, followed by an exploratory user study with five disabled voice coders to obtain their feedback on a fixed grammar voice coding *** provided new insights around the tools voice coders utilise, the need for multimodal coding approaches, as well as experiences in working within mixed ability development teams. Findings from the user evaluation elicited views around the need for distinct monosyllable voice commands, as well as other requirements to support efficient voice coding (e.g. command chaining, intuitive navigation, and custom command definition).Research limitations/implicationsThe research was conducted with experienced voice coders which may have influenced perceptions of the voice coding prototype. The system was also evaluated over a single evaluation session, whereas longitudinal research with novice voice coders could also present additional *** implicationsFurther research is required into customisable multimodal voice coding approaches for developers with physical impairments who may have varying levels of experience in coding via alternative ***/valueThe findings present new insights around the working practices and unique requirements of voice coders with physical impairments and highlights important new avenues for future research.

关键词： voice coding Accessibility Assistive technology Inclusive design voice interaction Disabled developers

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Low bit-rate voice compression based on frequency domain interpolative techniques

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IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2006年第2期14卷 558-576页

作者： Bhaskar, U Swarninathan, K Hughes Network Syst Germantown MD 20876 USA

This paper presents an approach, referred to as frequency domain interpolation (FDI), for achieving high-quality speech at low bit-rates (4 kb/s and below) within reasonable complexity and delay. FDI methods, like the prototype waveform interpolation (PWI) methods, derive a prototype waveform (PW) at regular intervals of time. But, unlike PWI, there is no separation into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW) component. Instead, the PW is encoded after gain normalization in magnitude-phase form. The magnitude is modeled as a sum of mean and deviation values in multiple frequency bands and this model is quantized using switched backward adaptive VQ techniques. The phase information is represented as a composite vector of PW correlations in multiple frequency bands and an overall voicing measure. This information is quantized using a VQ at the encoder. At the decoder, a phase model is employed that uses the received phase (and magnitude) information to reproduce PWs with the correct periodicity and evolutionary characteristics. Speech is synthesized by interpolating the reconstructed PWs after gain adjustment and filtering it using the short-term predictor and a postfilter. The design of a 4-kb/s and a 2.4-kb/s FDI codec are presented in this paper and their performance is characterized in terms of delay, complexity, and subjective voice quality. The results confirm that FDI techniques have the potential for delivering high-quality speech at low bit-rates in a cost-effective manner.

关键词： frequency domain interpolation (FDI) linear prediction prototype waveform interpolation voice coding

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Media coding for the next generation mobile system LTE

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COMPUTER COMMUNICATIONS 2010年第16期33卷 1916-1927页

作者： Jarvinen, Kari Bouazizi, Imed Laaksonen, Lasse Ojala, Pasi Ramo, Anssi Nokia Res Ctr Tampere 33720 Finland

Introduction of LTE (Long Term Evolution) brings enhanced quality for 3GPP multimedia services. The high throughput and low latency of LTE enable higher quality media coding than what is possible in UMTS. LIE-specific codecs have not yet been defined but work on them is ongoing in 3GPP. The LIE codecs are expected to improve the basic signal quality, but also to offer new capabilities such as extended audio bandwidth, stereo and multi-channels for voice and higher temporal and spatial resolutions for video. Due to the wide range of functionalities in media coding, LIE gives more flexibility for service provision to cope with heterogeneous terminal capabilities and transmission over heterogeneous network conditions. By adjusting the bit-rate, the computational complexity, and the spatial and temporal resolution of audio and video, transport and rendering can be optimised throughout the media path hence guaranteeing the best possible quality of service. (C) 2010 Elsevier B.V. All rights reserved.

关键词： Multimedia coding voice coding Audio coding Video coding

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A simulation study of adaptive voice communications on IP networks

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COMPUTER COMMUNICATIONS 2001年第9期24卷 757-767页

作者： Barberis, A Casetti, C De Martin, JC Meo, M Politecn Torino Dipartimento Elettron I-10129 Turin Italy Politecn Torino CNR IRITI Turin Italy

This paper presents simulation results outlining the behavior of rate-adaptive voice communications over IP networks. In the considered architecture, voice coders adapt their rate to the current state of the network so as to generate only the bandwidth that the network is capable of carrying. An algorithm is proposed for driving the transmission rate of voice sources on the basis of estimations of the network conditions, measured in terms of packet delays and losses. The effectiveness of the proposed solution is then investigated in various scenarios which comprise: (i) a dedicated network in which the available bandwidth is exclusively shared between adaptive voice connections: (ii) a scenario in which adaptive voice sources compete with other TCP-like sources;and (iii) an uncontrolled network environment. We have compared the performance of the rate-adaptive against the non-adaptive (i.e. fixed-rate) approach for the transport of voice over TP. Using a rate-adaptive approach, more voice communications can be carried while maintaining a good quality of service, even on non-segregated networks. (C) 2001 Elsevier Science B.V. All rights reserved.

关键词： IP telephony voice coding simulation

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Congestion avoidance using DYnamic COdec MAnagement: A solution for ISP

Congestion avoidance using DYnamic COdec MAnagement: A solut...

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11th Asia-Pacific Conference on Communications

作者： Alcuri, L Saitta, F Fasciana, ML Univ Palermo DIEET I-90128 Palermo Italy

ISBN: (纸本)0780391322

The large diffusion of VoIP is an essential key for the success of emerging Internet Service Providers. These ISPs conflict with the interests of historical and predominant network maintainers which often control the network infrastructure and telephone services. To win this competition, emerging ISPs have to adopt strategic QoS solutions which will capture the attention of network users and phone clients. This paper presents a solution which is able to obtain the same performance of a 64 Kbps channel reducing the costs for over-provisioning network bandwidth. DyCoMa is a mechanism studied for VoIP applications in network with a limited bandwidth for multimedia services. It works preventing network congestion without blocking incoming calls and maintaining under control end-to-end QoS. This solution could be adopted by ISP which would like to offer VoIP services without investing huge capital in bandwidth resources.

关键词： QoS voice over IP (VoIP) voice coding network simulator2 DyCoMa

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Multiple description and multi-path routing for robust voice transmission over ad hoc networks

Multiple description and multi-path routing for robust voice...

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2nd IEEE International Conference on Wireless and Mobile Computing, Networking and Communications (WiMob 2006)

作者： Kwong, M. D. Cherkaoui, S. Lefebvre, R. Univ Sherbrooke Dept Elect & Comp Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)1424404940

Achieving real-time voice communication over a mobile wireless ad hoc network poses many challenges both in speech coding and network protocols. In this paper, we investigate the real-time voice transmission capacity of ad hoc networks using a multiple description coding (MDC) scheme along with a new multi-path routing protocol called MSBR. The MDC generates two or more complementary bitstreams from the bitstream of the speech encoder, each sent along different routes. This allows the receiver to maintain acceptable speech quality even with missing packets. MSBR algorithm attempts to find multiple reliable stable routes to the destination node. Simulations using the GloMoSim network simulator and speech audio evaluation setups were used to study the performances of the approach in different network scenarios. Results show that the MDC scheme, together with MSBR, makes an effective use of the network and provides good performances for real time voice transmission.

关键词： voice coding multiple description multi-path routing ad hoc networks

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Simplified Sympes Codec with Positive DC Offset 24

Simplified Sympes Codec with Positive DC Offset

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24th IEEE International Conference on Electronics, Circuits and Systems (ICECS)

作者： Yarman, B. S. Korkmaz, O. Temizyurek, M. Ince, F. Hassoy, B. Canberi, H. Ceren, E. Gultekin, E. Istanbul Univ Istanbul Turkey RFT Co Teknopk Istanbul TR-34912 Istanbul Turkey STM Co Minist Def TR-06510 Ankara Turkey

ISBN: (纸本)9781538619117

In this paper, further improvement is reported in the simplified version of the Sympes voice encoding algorithm. A drastic reduction in the computation time of the algorithm is achieved by introducing an offset in the amplitude of the normalized speech signals which enables trusted, secure and real time voice communication over computer networks and internet.

关键词： Sympes voice coding Speech Modelling Stochastic Signals with Non-Zero Mean Principle Component Analysis Eigen Value Problem

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Speech Analysis for Processing of Musical Signals

Speech Analysis for Processing of Musical Signals

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作者： Mészáros, Tomáš Brno University of Technology

Hlavním cílem této práce je obohatit hudební signály charakteristikami lidské řeči. Práce zahrnuje tvorbu audioefektu inspirovaného efektem talk-box: analýzu hlasov... 详细信息

Hlavním cílem této práce je obohatit hudební signály charakteristikami lidské řeči. Práce zahrnuje tvorbu audioefektu inspirovaného efektem talk-box: analýzu hlasového ústrojí vhodným algoritmem jako je lineární predikce, a aplikaci odhadnutého filtru na hudební audio-signál. Důraz je kladen na dokonalou kvalitu výstupu, malou latenci a nízkou výpočetní náročnost pro použití v reálném čase. Výstupem práce je softwarový plugin využitelný v profesionálních aplikacích pro úpravu audia a při využití vhodné hardwarové platformy také pro živé hraní. Plugin emuluje reálné zařízení typu talk-box a poskytuje podobnou kvalitu výstupu s unikátním zvukem.

关键词： Lineární predikce Audio syntéza Kódování řeči Modelování zvukových efektů VST pluginy LADSPA DAW Zpracování signálů Linear prediction Audio synthesis voice coding Sound effects modeling VST plugins LADSPA DAW Signal processing Text

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Speech Compression

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INFORMATION 2016年第2期7卷 32-32页

作者： Gibson, Jerry D. Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93118 USA

Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed.

关键词： speech coding voice coding speech coding standards speech coding performance linear prediction of speech

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The Design and Implementation of a Network-Based Telephone Recording System

The Design and Implementation of a Network-Based Telephone R...

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20th Chinese Control and Decision Conference

作者： Gao Fu-xiang Wang Xin Li Feng-yun Northeastern Univ Coll Informat Sci & Engn Shenyang 110004 Peoples R China

ISBN: (纸本)9781424417339

With the growth of the demand for voice recording in the fields of railway and electric power, many kinds of digital voice recorders have appeared. General voice recorders can only record voice produced at one place, when voice appeared in many different places need to be recorded centrally, these kinds of recorders become helpless. The problem can be solved with a kind of network-based voice recording system. It uses microcontroller, voice encoder and Ethernet controller as the core components, and is programmed with the TCP/IP protocol stack. It can be used to transmit the voice data to a remote server for storage. Compared with the traditional recorder, it is cheaper, more flexible, and easier to update.

关键词： Embedded Internet Ethernet TCP/IP protocol stack Telephone recording voice coding

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